| 00:00.08 | carrar | dman phone |
| 00:00.09 | carrar | brb |
| 00:00.20 | hardwire | grr |
| 00:00.24 | hardwire | chan_zap.c: We're Zap/53-1, not Zap/53-1<ZOMBIE> |
| 00:00.32 | hardwire | wot! does! it! mean! |
| 00:00.37 | hardwire | and why does it make my asterisk go poof |
| 00:01.11 | bmoraca | we're looking in to bonded SDSL (3mb both ways), and then using a GRE tunnel to get our IP addresses here...but getting the bossman to make a decision about money is tough |
| 00:03.05 | bmoraca | which is weird, because it would save about $1000/mo in the end to do it this way...because then we could send our voice traffic over the internet to our colocation trunk group which we use to sell VoIP service...allowing us to get rid of our PRI at our office as well as our point-to-point T1. Plus the rates are WAAAY better. |
| 00:08.37 | *** part/#asterisk jm|home (n=jm|home@dilbert.jamiem.com) |
| 00:09.30 | *** part/#asterisk Deeewayne (n=dwayne@nat/digium/x-a7816f5ef2f0333c) |
| 00:09.35 | hardwire | I found my zombie problem |
| 00:09.36 | hardwire | brainz |
| 00:09.50 | hardwire | I was using Dial(...,g) then going to Wait(10) |
| 00:09.55 | hardwire | right after that |
| 00:10.16 | hardwire | because the people at the call center I'm working with freaked out that their channel bank on span 1 was beeping at them when people hung up on them. |
| 00:10.32 | hardwire | well.. their analog phones on the channel bank tied to span 1 on the asterisk box |
| 00:10.38 | hardwire | so I made it wait(10) after the dial |
| 00:10.43 | hardwire | and it's causing all sorts of issues. |
| 00:11.57 | *** join/#asterisk fskrotzki (n=fskrotzk@rrcs-208-125-76-233.nys.biz.rr.com) |
| 00:12.18 | *** join/#asterisk neurosys (n=vinix@c-71-196-19-254.hsd1.fl.comcast.net) |
| 00:12.25 | hardwire | will there typically be issues with continuing on the dialplan after a dial for zap callers? |
| 00:12.33 | hardwire | this is the first time I have experienced this |
| 00:13.32 | hardwire | and will boris and natasha ever capture bullwinkle and that crazy flying squirrel? |
| 00:14.12 | *** join/#asterisk ingenius (n=alektro@host251.190-31-44.telecom.net.ar) |
| 00:14.31 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 00:15.04 | *** join/#asterisk stevetotaro (n=Steve@pool-72-72-143-197.hrbgpa.dsl-w.verizon.net) |
| 00:17.52 | ManxPower | hardwire: Playback(silence) for 10 seconds instead of a wait see if that helps |
| 00:21.47 | hardwire | ManxPower: that's what I suggested to the client |
| 00:22.47 | hardwire | ManxPower: I didn't disable three way calling either |
| 00:23.04 | bmoraca | hardwire: disable all calling and then there won't be any problems at all! |
| 00:23.13 | hardwire | and now that I know that "Zombie" is the right word for what is happening, google has helped me discover that threewaycalling makes issues happen |
| 00:23.28 | hardwire | and it shouldn't have been enabled in the first place |
| 00:23.32 | hardwire | I just skipped over it |
| 00:23.34 | hardwire | nuhr |
| 00:23.55 | bmoraca | why does everyone make that sound? |
| 00:24.00 | bmoraca | people here do it ALL THE TIME |
| 00:26.56 | *** join/#asterisk erth64net (n=erth64ne@69-30-67-191.dq1sn.easystreet.com) |
| 00:29.15 | hardwire | bmoraca: like when they talk to you they say "nuhr"? |
| 00:29.19 | hardwire | I'd take that as a sign. |
| 00:32.42 | *** join/#asterisk Bonix (n=Bonix@200-195-41-212.isimples.com.br) |
| 00:36.44 | *** join/#asterisk erth64net (n=erth64ne@69-30-67-191.dq1sn.easystreet.com) |
| 00:43.53 | bmoraca | fuckin' ay. my monitor just died :( |
| 00:47.40 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 00:52.26 | harry_v | is it crt? |
| 00:52.53 | harry_v | I have a older Flexscan high graphics 20 inch that is flickering off and on. probebly last legs. |
| 00:55.13 | *** join/#asterisk killown (n=Yamato@unaffiliated/killown) |
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| 01:00.38 | *** join/#asterisk keith4_ (n=kbe2@216-164-144-176.c3-0.eas-ubr9.atw-eas.pa.cable.rcn.com) |
| 01:01.27 | keith4_ | i'm in a meetme channel with a handful of other SIP users. one jerk on a cell phone joins, and suddenly i'm hearing echo whenever anyone talks. can this be fixed? |
| 01:02.35 | jaytee | tell the jerk with the cellphone to hangup |
| 01:03.44 | *** join/#asterisk dwayne (n=dwayne@c-76-29-245-9.hsd1.al.comcast.net) |
| 01:04.35 | JJx3 | for an assistance channel ppl aint that helpful |
| 01:05.01 | keith4_ | parts per liter? |
| 01:05.24 | *** join/#asterisk killown (n=Yamato@unaffiliated/killown) |
| 01:05.26 | keith4_ | for someone in the country that invented English, your grasp of it ain't that strong |
| 01:05.30 | JJx3 | thanks for making my point keith4 |
| 01:05.49 | JJx3 | i'm not English |
| 01:06.21 | JJx3 | typical freaking yank |
| 01:06.48 | keith4_ | the channel is much helpful earlier in the day |
| 01:06.54 | keith4_ | much *more* helpful, rather |
| 01:07.02 | jaytee | omg! did he say "yank"? |
| 01:07.18 | JJx3 | and you're having a go @ ne for "my" english? LMFAO |
| 01:07.38 | JJx3 | sorry jaytee, i meant WANK! |
| 01:07.43 | keith4_ | i didn't say you were English |
| 01:08.06 | JJx3 | you said "<keith4_> for someone in the country that invented English, your grasp of it ain't that strong" |
| 01:08.13 | keith4_ | very good |
| 01:08.19 | JJx3 | England = English language |
| 01:08.28 | keith4_ | oh my |
| 01:08.30 | keith4_ | you are confused |
| 01:08.47 | JJx3 | no, you're just a tossbag |
| 01:09.26 | *** join/#asterisk ruben23 (n=IT-ADMIN@124.107.3.178) |
| 01:09.47 | keith4_ | i'm not the one who is criticizing people while simultaneously asking them for help |
| 01:09.52 | ruben23 | hi |
| 01:10.18 | JJx3 | did i ask you for help keith4? |
| 01:10.27 | jaytee | JJx3, expecting some high level of help response in here is setting your standards too high. no one in this room gets paid to be here. If someone doesn't answer your question right away try asking a little while later. If no one answers then maybe it's because the question is too basic and shows you haven't "done your own homework" or aren't willing to put the time into reading the available help materials. |
| 01:12.01 | JJx3 | jaytee, i was talking to one of the @'s about my issue earlier, and even tho helpful, he had to leave & since then all I have got is BS |
| 01:12.27 | JJx3 | however, thanyou for your lenghly explanation of the channel |
| 01:13.43 | jaytee | glad to be of help! :-) |
| 01:14.34 | JJx3 | jaytee, may i ask you, would * on ubuntu server 8.10 be better than *NOW 1.2 ? |
| 01:14.57 | *** join/#asterisk askmeagain (n=ask@modemcable147.212-70-69.mc.videotron.ca) |
| 01:15.00 | askmeagain | Would anyone be kind enough to give me a hand with customizing extensions_custom.conf on my trixbox so I can hang up the line for emergency call ? I tried with the example in the PDF TRixbox without tears @ page 161 with no success |
| 01:15.08 | jaytee | JJx3, not in my humble opinion |
| 01:15.15 | jaytee | care to hear my reasoning? |
| 01:15.21 | JJx3 | ok kool |
| 01:15.36 | jaytee | JJx3, just a second |
| 01:15.46 | JJx3 | jaytee, do you know anything about polarity reversal in the UK ? |
| 01:15.48 | JJx3 | kool |
| 01:15.48 | jaytee | askmeagain, this is not the #trixbox channel |
| 01:16.08 | jaytee | JJx3, nope I'm in the US or as you'd say, a Yank |
| 01:16.22 | askmeagain | I was thinking it was based on asterisk and tehrefore be the same |
| 01:16.24 | keith4_ | ubuntu server? ugh |
| 01:16.28 | keith4_ | askmeagain: nope |
| 01:16.32 | askmeagain | since it's not done through the gui |
| 01:16.52 | JJx3 | keith4, if i wanted to hear from an ass, I would have farted |
| 01:17.23 | keith4_ | lucky for me, i don't care what you want |
| 01:17.24 | jaytee | my reasons for avoiding Ubu 8.10 is that it's TOO bleeding edge. It's better to use a slightly less "edgy" kernel and base system for stability and to avoid incompatibility issues. |
| 01:18.38 | *** join/#asterisk neurosys (n=vinix@c-71-196-19-254.hsd1.fl.comcast.net) |
| 01:19.06 | voxter | anyone here used a polycom ip6000? |
| 01:19.09 | harry_v | keith, you use ubunto? |
| 01:19.43 | JJx3 | jaytee, what would you rec'mend for a very basic * system, 2 - 3 extensions ? |
| 01:19.57 | harry_v | I am all for stability. a tech starts to sweat when the customer is unhappy something is taking to long to to get fixed. |
| 01:20.04 | harry_v | 6000? |
| 01:20.34 | keith4_ | harry_v: maybe on a laptop. never on a server |
| 01:21.39 | harry_v | btw, a employment office did not have its windows licences for a crashed windows 2000 install, so tried to install ubunto but it hit a wall with some kind of swap error. nothing I have seen before. |
| 01:22.00 | harry_v | rather use the command to do the install. |
| 01:22.00 | jaytee | if you like Ubuntu as a distro then I'd recommend the current stable version of Debian, it's similar enough in the way it varies from the LSB like Ubuntu. Myself, I prefer RHEL 5.x or CentOS 5.x |
| 01:22.35 | harry_v | yea if it can install on a old Compaq en pro work station with scsi on it. |
| 01:22.42 | JJx3 | cool, i might kill my existing * server for RHEL |
| 01:23.16 | harry_v | I really liked using fedora. everything installed so smoothly on it. Use centos currently |
| 01:23.33 | jaytee | for a very basic system all you need is a decent P3 or P4 with 512MB of RAM, CentOS 5.2 and Asterisk 1.4.x |
| 01:23.43 | voxter | This ip6000 conference phone powers up via poe, but not if i plug it in with the ac adapter... very strange |
| 01:24.16 | jaytee | and you could probably get by on less ram but if you do it means you're a cheap bastard and should stay out of the voip business before you give the rest of us a bad name. |
| 01:25.48 | harry_v | jaytee, you talking to me? |
| 01:26.12 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 01:26.36 | jaytee | harry_v, no. JJx3 asked me what I'd rec'mend for very basic * system, 2 - 3 extensions |
| 01:29.42 | harry_v | jaytee, you wouldnt know of a dial plan that can enable my CID for one time calls? |
| 01:30.32 | jaytee | enable your CID for one time calls? Could you be more specific in your description, I'm not sure I understand what you're trying to accomplish? |
| 01:30.47 | *** join/#asterisk pa (n=pa@unaffiliated/pa) |
| 01:31.18 | keith4_ | harry_v: you want to use a vertical service code to enable CID on-demand? |
| 01:31.26 | *** join/#asterisk erth64net (n=erth64ne@69-30-67-191.dq1sn.easystreet.com) |
| 01:31.47 | harry_v | well, by default we do not have CID. now you can press I think *82 to enable CID so calls can go though to those parties who reject numbers without CID enabled. |
| 01:32.28 | harry_v | I have tried varios combinations of code to no avail. I no longer have them. |
| 01:32.56 | *** join/#asterisk Kumbang (n=unknown@125.163.83.153) |
| 01:33.37 | jaytee | harry_v, are we talking about calls to the PSTN? |
| 01:33.41 | harry_v | yes |
| 01:33.58 | jaytee | and most of the time you don't want to have CID enabled, correct? |
| 01:34.13 | harry_v | thats my wife me I dont care but its under her name :) |
| 01:34.37 | harry_v | So there are times, it neeeds to be enabled on a call to call basis. |
| 01:34.48 | jaytee | do you dial a 9 for an outside line? |
| 01:34.56 | harry_v | no |
| 01:35.39 | *** join/#asterisk killown (n=Yamato@unaffiliated/killown) |
| 01:36.41 | harry_v | what else do you need to know? |
| 01:38.14 | jaytee | harry_v, why not modify your outbound context to include pattern matches, one for local and one for long-distance starting with a 9 then and have the first priority in each pattern do exten => _9NXXXXXX,1,Set(CALLERID(num)="yournumbergoeshere") |
| 01:38.43 | jaytee | then when you want to dial an outside number and show callerid you just dial a 9 first |
| 01:39.50 | *** join/#asterisk dgoner (n=david@mx1.repairpc.net) |
| 01:40.51 | harry_v | I see |
| 01:40.55 | *** join/#asterisk MaliutaLap (n=biteme@kiev.lusan.id.au) |
| 01:41.01 | harry_v | Icould do that |
| 01:41.31 | harry_v | but |
| 01:41.36 | harry_v | I dont think that will work |
| 01:43.03 | harry_v | when you press *82, the dial tone will pulse for about 2-3 seconds before you get a steady dial tone. your option will not work because there needs to be a pause before it will dial the rest of the number after dilaing *82 |
| 01:43.34 | keith4_ | so, you want to be able to dial 9 first, maybe, and have that perform the *82 for you |
| 01:43.40 | keith4_ | and then dial your number as usual |
| 01:44.22 | jaytee | harry_v, you're using analog lines with a digium or sangoma FXO card? |
| 01:44.53 | harry_v | digium |
| 01:45.15 | harry_v | actually, *82 can do the same as 9 |
| 01:46.34 | harry_v | anyone here who will use our phones will not know that you need to dial 9. best to substitute with *8X as per instruction by the phone company. |
| 01:47.14 | keith4_ | so you just want your dialplan to support dialing *8X before 10-digit dialing? |
| 01:47.19 | *** join/#asterisk MaliutaLap (n=biteme@203.171.192.31) |
| 01:47.27 | jaytee | exten => _NXXXXXX,1,Dial(Zap/1/www*82,15,mD(${EXTEN})) would do what you want for an analog zap channel dialing a local 7 digit number |
| 01:48.50 | harry_v | okay |
| 01:49.27 | jaytee | or you could modify it so it would be exten => _*82NXXXXXX,1,Dial(Zap/1/www*82,15,mD(${EXTEN})) |
| 01:49.35 | harry_v | I was going to say |
| 01:49.35 | harry_v | ;) |
| 01:50.07 | jaytee | and add another pattern match in your outbound context in the dialplan for *83 or whatever turns off CID at the PSTN |
| 01:50.22 | harry_v | might want to leave a playback message of dial that combined number if the caller dials just *82 |
| 01:52.01 | jaytee | harry_v, * is the Erector set of hybrid PBX systems, you can pretty much build it if you can imagine it :-) |
| 01:52.15 | harry_v | I know |
| 01:52.25 | harry_v | :) |
| 01:52.40 | harry_v | its also what partially killed nortel |
| 01:53.04 | jaytee | harry_v, in a little over a week I'm pulling the plug on our Option 11C |
| 01:53.20 | harry_v | Isee |
| 01:53.26 | jaytee | I've migrated 127 phones and only have a few left and the CAP to move over |
| 01:54.50 | carrar | You are gonna miss that carrier-grade, five-nines reliability |
| 01:55.04 | carrar | and those sexy M2008 handsets |
| 01:55.54 | carrar | looking at some add for the 11c |
| 01:55.56 | carrar | ad |
| 01:55.57 | jaytee | "Um, you're going to need licenses for more DN's and if you want CID on the analog phones you'll need this CLASS modem card with licenses, it's $3700.00 and you should increase your VM storage. We can unlock another 4GB of storage on the 80GB drive on the CallPilot server for another 3000.00." "No, I've got a better idea. Fuck you!" |
| 01:55.59 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 01:56.22 | carrar | hahah |
| 01:57.44 | jaytee | seriously, the story goes that Mark wanted a PBX for his first business before it became Digium and it just did Linux support but the pricing was so ridiculously expensive he said, "Screw it! I'll write my own software and engineer my own cards." Voila! Asterisk was born. |
| 01:58.19 | jaytee | Thank you, Nortel! Thank you, Avaya! You're pricing schemes have become your own undoing. |
| 01:58.35 | carrar | tell us how you really feel |
| 01:58.43 | trnzmeta | yes maintenance contracts are way expensive |
| 01:58.57 | carrar | everything is way too expensive |
| 01:59.02 | carrar | even more nowdays |
| 02:00.20 | *** part/#asterisk ruben23 (n=IT-ADMIN@124.107.3.178) |
| 02:00.58 | harry_v | jaytee, take a look at this. did not work |
| 02:01.02 | harry_v | http://www.pastebin.ca/1342465 |
| 02:02.07 | jaytee | Sending DTMF {$EXTEN} <<<< DUH!!!! you've got a typo |
| 02:02.10 | jaytee | do you see it? |
| 02:02.11 | *** join/#asterisk killown (n=Yamato@unaffiliated/killown) |
| 02:02.22 | carrar | ${ |
| 02:02.37 | carrar | err sorry |
| 02:02.52 | jaytee | exten => _NXXXXXX,1,Dial(Zap/1/www*82,15,mD(${EXTEN})) |
| 02:03.00 | carrar | taking all the fun out of learning |
| 02:03.47 | jaytee | carrar, could you come over to my place tomorrow night and spoon feed me ice cream so that I don't have to use my own hands? Maybe do my dishes and vacuum while you're here? |
| 02:03.58 | carrar | yes I can do it all |
| 02:04.08 | carrar | I have nothing else to do |
| 02:04.22 | harry_v | oops |
| 02:04.23 | harry_v | :) |
| 02:04.27 | jaytee | carrar, has anyone ever told you you're easy :-) |
| 02:04.28 | carrar | Would you like me to bring some strippers too? |
| 02:05.02 | carrar | surely you don't want to watch me work |
| 02:05.41 | jaytee | no, actually that would be the most fun part of it all |
| 02:06.08 | jaytee | drmessano, hey! are you here? |
| 02:06.20 | *** join/#asterisk stuffcorpse (n=rick@121.98.136.241) |
| 02:07.45 | carrar | hrmm |
| 02:07.53 | carrar | I should eat some dinner |
| 02:08.05 | carrar | I should twitter that out |
| 02:08.13 | carrar | oh wait, I don't use twitter |
| 02:08.23 | harry_v | - Sending DTMF '*82604xxxxxxx' to the called party. |
| 02:08.23 | harry_v | <PROTECTED> |
| 02:08.38 | harry_v | hung up after sending all the numbers including *82 |
| 02:08.52 | carrar | xxx are valid numbers!! :) |
| 02:08.55 | carrar | are not |
| 02:09.09 | harry_v | Im not going to display my phone number here |
| 02:09.13 | carrar | haha |
| 02:09.17 | carrar | wuss |
| 02:09.21 | harry_v | :) |
| 02:09.33 | carrar | I have a fax machine here waiting to call you |
| 02:09.43 | harry_v | ohh im sure :) |
| 02:09.48 | jaytee | harry_v, I'm sorry. I think I screwed up |
| 02:10.11 | keith4_ | i thought you wanted to pause between *82 and the 10-digit |
| 02:10.11 | harry_v | its a tricky dial plan. not to much responce from asterisk.org ;) |
| 02:10.20 | jaytee | pastebin your extensions.conf and mask the actual number with nxxxxxx but leave everything else the same, ok? |
| 02:10.49 | harry_v | yes, that is how telus dial tone works. you dial *82, it pulses 2-3 times then you dial the 10 digits. |
| 02:11.01 | carrar | haha |
| 02:11.07 | carrar | how un fun |
| 02:11.50 | harry_v | well, I never really did it in one continious sequence but I suspect thats how it works andalways has |
| 02:12.19 | carrar | good thing I missed the part where you eplain what you are trying to do |
| 02:12.55 | carrar | goes back to spoon feeding jaytee ice cream |
| 02:13.24 | jaytee | harry_v, if your dial line looks like this, exten => _*82NXXXXXXXXX,1,Dial(Zap/2/www*82,15,mD(${EXTEN})) then you actually need it to look like this exten => _*82NXXXXXXXXX,1,Dial(Zap/2/www*82,15,mD(${EXTEN:3})) |
| 02:13.27 | harry_v | http://www.pastebin.ca/1342474 |
| 02:13.48 | harry_v | so add :3 |
| 02:14.00 | harry_v | im going out 1 |
| 02:14.02 | harry_v | :) |
| 02:14.03 | jaytee | yeah, that'll strip the *82 out of the exten. |
| 02:14.21 | *** join/#asterisk double_cheesburg (n=chatzill@ip70-162-238-155.ph.ph.cox.net) |
| 02:14.26 | jaytee | our you could change the *82 in the pattern mask to just be 9 and then do a :1 instead |
| 02:14.35 | carrar | just get pri and set the privacy! :) |
| 02:14.42 | jaytee | saves having to dial 2 extra digits, you only dial 1 extra digit |
| 02:14.53 | harry_v | worked! |
| 02:15.01 | jaytee | carrar, yeah! everyone should have their own PRI!!! |
| 02:15.01 | harry_v | wonderfull |
| 02:15.41 | harry_v | nice nice nice. thankyou very much. |
| 02:15.54 | jaytee | harry_v, cool! so you can figure out how to modify that to suit your tastes, needs I'm sure |
| 02:16.13 | jaytee | god, I love this movie |
| 02:16.30 | harry_v | I should add music on hold for all extentions. people generally dont like IVRs so at least the music would calm them a bit ;) |
| 02:16.51 | harry_v | yea, I need to make my own crazy dial plan howtos. |
| 02:16.56 | jaytee | that's what the lower case m does in front of the D |
| 02:18.06 | harry_v | Some one interfaces asterisk with a linux CCTV application. Any time there was movment at a particualer camera it would call the owner which camera was sensing motion. |
| 02:18.48 | jaytee | that's nothing, [TK]D-Fender setup his * box to turn on the coffee pot and call him when the coffee was ready. |
| 02:18.56 | harry_v | hahah |
| 02:19.28 | jaytee | One guy uses X10 technology on his * box to call from work to adjust the heating/AC zones in his house or set his TIVO to record something |
| 02:19.33 | harry_v | that is pretty good :) |
| 02:19.46 | harry_v | I was just going to say did use a x10 |
| 02:19.49 | harry_v | ? |
| 02:20.39 | jaytee | yeah, there's OSS out there for X10 on linux |
| 02:21.16 | jaytee | use some bash scripts called by System() and you've got it made in the shade |
| 02:21.39 | harry_v | that is something that I am gradually learning. |
| 02:22.45 | hardwire | so in newer polycoms the firmware release is all done through one sip.ld? |
| 02:23.06 | harry_v | here is something I do want to do. Hook up a high pressure house and camera to a XYZ stepper motor and peg tagers as thay spray paint on a wall. |
| 02:23.11 | hardwire | one.. 38 meg.. file? |
| 02:23.26 | hardwire | ah |
| 02:23.27 | hardwire | the split! |
| 02:23.30 | hardwire | kisses the sky |
| 02:23.35 | hardwire | this should make me happy |
| 02:23.43 | jaytee | hardwire, no they split the sip.ld files up by model |
| 02:23.48 | hardwire | I found it |
| 02:23.49 | hardwire | sigh |
| 02:23.58 | hardwire | what did I say about ignoring me earlier? |
| 02:24.06 | jaytee | oh, sorry! forgot. |
| 02:24.20 | hardwire | you'll regret this. |
| 02:24.21 | jaytee | puts hardwire back on ignore :-) |
| 02:24.47 | hardwire | I stole a picnic sized bag of doritos from a bbq earlier |
| 02:24.49 | hardwire | I'm so happy |
| 02:24.58 | hardwire | my cat would be in heaven |
| 02:25.04 | hardwire | crawlin in the bag-n-stuff |
| 02:26.00 | hardwire | http://photos-a.ll.facebook.com/photos-ll-snc1/v1948/173/74/1013627307/n1013627307_30274616_7790.jpg |
| 02:27.26 | jaytee | awwww |
| 02:27.38 | *** join/#asterisk umpc (n=Justin@unaffiliated/umpc) |
| 02:27.39 | jaytee | what's the cat's name |
| 02:27.42 | hardwire | Miachu. |
| 02:27.57 | jaytee | what's that mean? |
| 02:27.57 | hardwire | The Destroyer. |
| 02:28.04 | hardwire | Mia (her name) chu (cause I'm a freak) |
| 02:28.04 | jaytee | hmmm |
| 02:28.29 | hardwire | she has dug to the bottom of a full trash can to find old taco bell. |
| 02:28.38 | hardwire | her scrounging skills are unmatched. |
| 02:30.31 | harry_v | I might do a x10 for my wife pot roast ;) |
| 02:31.38 | harry_v | have it call her and ask if she would like to set the time to start cooking a hour before she leaves ;) |
| 02:33.20 | *** join/#asterisk russellb_ (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 02:33.20 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 02:33.40 | hardwire | 0101001556|copy |3|00|Download of '2345-12500-001.sip.ld' succeeded on attempt 1 (addr 1 of 1) |
| 02:33.41 | hardwire | 0101001556|app1 |4|00|Application 2345-12500-001.sip.ld is not is incompatible with the phone. |
| 02:33.41 | hardwire | 0101001605|app1 |4|00|Loaded application sip.ld successfully, errors 0x120. |
| 02:33.46 | hardwire | I.. Love!!! Double Negatives!! |
| 02:33.49 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 02:36.56 | jaytee | IP330 |
| 02:37.16 | hardwire | its a 550 |
| 02:37.20 | hardwire | why don't it know betta? |
| 02:37.32 | jaytee | maybe it does |
| 02:38.31 | hardwire | mebbe |
| 02:39.03 | jaytee | yeah, I got it wrong, it 2345-12200-000.sip.ld for the 330 |
| 02:39.15 | hardwire | grr. |
| 02:39.21 | jaytee | 12500 is the right one for a 550 |
| 02:39.27 | *** join/#asterisk killown (n=Yamato@unaffiliated/killown) |
| 02:39.46 | *** part/#asterisk killown (n=Yamato@unaffiliated/killown) |
| 02:40.45 | *** join/#asterisk Maliuta (n=scooby@kiev.lusan.id.au) |
| 03:07.18 | *** join/#asterisk rpm (n=rpm@S010600055d2cf2e2.cg.shawcable.net) |
| 03:07.43 | harry_v | talking about polycoms would not mind enabeling some of my lines, some more soft buttons and have it load a bmp company logo on the front of the phone. |
| 03:14.41 | hardwire | jaytee: downloaded the bootrom updates.. it's all kosher now |
| 03:15.06 | jaytee | yay!!!! |
| 03:17.34 | hardwire | yay indeed little buddy. |
| 03:23.37 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 03:23.48 | jaytee | little buddy? now I'm feeling like Gilligan |
| 03:25.13 | eppigy | hello |
| 03:25.20 | jaytee | you are dave |
| 03:25.25 | eppigy | yes sir |
| 03:29.51 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 03:31.25 | *** join/#asterisk ruben23 (n=IT-ADMIN@124.107.3.178) |
| 03:31.34 | *** join/#asterisk freakazoid0223 (n=matt@pool-71-246-17-74.phlapa.fios.verizon.net) |
| 03:35.54 | *** join/#asterisk thehar (i=thehar@thehar.xmission.com) |
| 03:39.34 | thehar | herro |
| 03:41.04 | drmessano | PIKACHU |
| 03:44.06 | thehar | will polycom ip 650s play well with asterisk? |
| 03:44.10 | drmessano | ~ihazbox |
| 03:44.11 | thehar | i have 600s and 601s |
| 03:44.14 | drmessano | ~ihaznew |
| 03:44.18 | thehar | and they are so much purtier as new replacements |
| 03:45.13 | drmessano | jaytee |
| 03:45.44 | jaytee | drmessano |
| 03:48.27 | russellb_ | i have a couple of 650s on my desk |
| 03:48.30 | russellb_ | they work just fine and dandy |
| 03:49.03 | drmessano | ~howcaninstall |
| 03:49.04 | jbot | http://mine.icanhascheezburger.com/view.aspx?ciid=3455370 |
| 03:49.06 | drmessano | ^^^^^^^^^^ |
| 03:49.11 | *** join/#asterisk eric256 (n=eric256@c-75-71-188-88.hsd1.co.comcast.net) |
| 03:50.08 | eric256 | is there anyway at the command line to get a list of channels? i tried asterisk -rx "show channels" and it works...but it cuts off the channel because of its fixed width display |
| 03:50.20 | thehar | russellb_: yay ty for kindly answering mes |
| 03:50.20 | russellb_ | hi. |
| 03:50.27 | russellb_ | thehar: np |
| 03:50.38 | thehar | russellb_: wat about that special codecs for it. |
| 03:50.49 | thehar | "hd" *snickers |
| 03:50.51 | thehar | g.blah |
| 03:50.58 | russellb_ | thehar: asterisk 1.4 supports passthrough/recording/playback |
| 03:51.02 | russellb_ | asterisk 1.6 supports transcoding |
| 03:51.27 | thehar | move 1.6 from trunk to stable pls |
| 03:51.38 | eric256 | i tried "show channels verbose" but it still cuts off the channels..... |
| 03:51.41 | russellb_ | i'd backport it for you for a cookie |
| 03:52.03 | thehar | what about a hug from mark? |
| 03:52.07 | thehar | or aaron |
| 03:52.11 | russellb_ | no dice |
| 03:52.12 | thehar | or tilghman |
| 03:52.37 | thehar | a fudgey nut basket? |
| 03:53.00 | russellb_ | heh, nah, i was just kidding ... |
| 03:53.02 | thehar | hehehe |
| 03:53.14 | thehar | what floor are you on? or will i have to ask someone else |
| 03:53.17 | russellb_ | should be easy to backport |
| 03:53.18 | *** join/#asterisk rpm (n=rpm@S010600055d2cf2e2.cg.shawcable.net) |
| 03:53.21 | thehar | orly? +D |
| 03:53.25 | russellb_ | really, i don't need anything |
| 03:53.28 | thehar | hehe |
| 03:53.46 | drmessano | or you could just actually use 1.6 |
| 03:53.51 | russellb_ | just drop me an email sometime to remind me and i'll do it when i'm bored |
| 03:53.59 | thehar | sweet. it's not time pressing |
| 03:54.04 | drmessano | Which is incredibly stable |
| 03:54.15 | thehar | but if i can tell my vp of ops that i can give him "HD VOICE" then he'll front the extra $40 per phone |
| 03:54.33 | Kyosh | hd voice aint all that great |
| 03:54.42 | eric256 | channels? anyone...i'll give out cookies...or anything |
| 03:54.43 | eric256 | lol |
| 03:54.56 | russellb_ | eric256: you're better off using the manager interface for that |
| 03:54.56 | thehar | i hate buzz words |
| 03:54.57 | eric256 | hd voice just means people here more than they are used to and think other phones are broken ;) |
| 03:55.06 | drmessano | I cant imagine G722 being all that compelling over the same limited phone handset |
| 03:55.13 | russellb_ | eric256: but i think there is a "show channels concise" command that some people use |
| 03:55.19 | drmessano | Maybe for conferencing on real conference sets |
| 03:55.19 | eric256 | i tried using Asterisk::Manager to "show channels" but it doesn't display them all |
| 03:55.33 | Kyosh | according to polycom, their hd voice works best with someone using hd voice on the other side |
| 03:55.46 | drmessano | lol |
| 03:55.51 | russellb_ | well sure, otherwise you don't gain anything. |
| 03:55.54 | thehar | WEIRD |
| 03:55.58 | russellb_ | and it really does sound amazing |
| 03:56.03 | thehar | you don't say |
| 03:56.04 | russellb_ | allison in wideband <3 |
| 03:56.04 | Kyosh | it does |
| 03:56.05 | thehar | snickers |
| 03:56.09 | thehar | oh allison |
| 03:56.26 | eric256 | concise...thanks thats perfect, based on the description i thought the default view was concise! |
| 03:56.28 | eric256 | ;) |
| 03:56.34 | russellb_ | eric256: heh, nah |
| 03:56.39 | russellb_ | eric256: glad that works for oyu |
| 03:57.04 | eric256 | yea i just needed the channel so i could steal calls from a custom HTML gui...kinda hack till i find something better |
| 03:57.16 | *** join/#asterisk MaliutaLap (n=biteme@203.171.195.179) |
| 03:57.23 | *** join/#asterisk jpeeler-home (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 04:01.04 | eric256 | now to figure out what these all are ;) actualy i don't need all of them anyway |
| 04:05.08 | bkw_ | Kyosh: don't let polycom fool you.. G722 has been around for ages.. you can even do it over the PSTN if both end support the capability |
| 04:05.37 | robwafle | hello? |
| 04:05.40 | bkw_ | drmessano: wideband conferences really lower the stress level of longer conferences |
| 04:05.54 | bkw_ | you spend less brain power trying to decode what everyone is saying |
| 04:06.27 | robwafle | bkw_: huh, what? :) |
| 04:06.41 | bkw_ | robwafle: talking about wideband conferences |
| 04:06.59 | robwafle | oh wait wide band probably means "higher quality" ? |
| 04:07.04 | bkw_ | right |
| 04:07.07 | robwafle | luke G711u? |
| 04:07.14 | robwafle | like, rather. |
| 04:07.15 | bkw_ | 8kHz = narrow band |
| 04:07.15 | thehar | don't you have a channel of your own? |
| 04:07.18 | thehar | snickers at bkw_ |
| 04:07.25 | thehar | ducks |
| 04:07.27 | robwafle | ah hah |
| 04:07.33 | robwafle | yeah that could be painful |
| 04:07.36 | bkw_ | 16, 32 and even 48kHz |
| 04:07.41 | bkw_ | thehar: hey guy |
| 04:07.48 | thehar | lolz |
| 04:08.02 | harry_v | bkw is right, brain uses less oxygen when its thinking in simple mode. |
| 04:08.48 | *** join/#asterisk MindTheGap (n=MindTheG@189.59.202.145) |
| 04:08.50 | robwafle | lol, more cpu power used, less brain power used.. I'm all about that. |
| 04:08.50 | bkw_ | plus it just sounds better |
| 04:08.58 | bkw_ | robwafle: it doesn't use more CPU |
| 04:09.00 | bkw_ | or more bandwidth |
| 04:09.06 | robwafle | oh yeah? |
| 04:09.08 | bkw_ | yep |
| 04:09.08 | harry_v | 3d PET scans shows some regions of the brain illuminated in red=more consumption of oxygen vs blue=much less consumption when not solving a complex puzzle. |
| 04:09.09 | harry_v | ;) |
| 04:09.15 | bkw_ | G722 is 64k just like ulaw |
| 04:09.31 | harry_v | ohh cpu talk |
| 04:09.39 | bkw_ | and the complexity is low |
| 04:09.46 | bkw_ | even G722.1 isn't very complex |
| 04:09.56 | bkw_ | and that uses even less bandwidth |
| 04:10.14 | harry_v | came across a full version of linux with 3d paging on a floppy. It was written in asm |
| 04:10.22 | drmessano | bkw_: Sure, but using the same $.99 handset (ok, $2 for a good phone), with the same 1/2 inch speaker element isn't that compelling.. Conference boxes are the only place you could even explore noticable audio improvement |
| 04:10.22 | harry_v | blew me away. |
| 04:11.16 | bkw_ | drmessano: even shitty phones get an improvement |
| 04:11.20 | robwafle | G722.1 < 64k ? |
| 04:11.31 | bkw_ | robwafle: it can run at 24, 32 and 48kbit |
| 04:11.32 | thehar | 48, 56, 64 |
| 04:11.36 | robwafle | ah hah |
| 04:11.43 | robwafle | i have a PIII 500Mhx |
| 04:11.48 | robwafle | Mhz |
| 04:11.49 | robwafle | i can't type |
| 04:11.54 | robwafle | (if you hadn't noticed..) |
| 04:11.56 | bkw_ | thats ok .. I can never type |
| 04:14.01 | robwafle | what version of asterisk do I need to use G722.1? |
| 04:14.20 | bkw_ | asterisk can't do g722.1 yet |
| 04:14.29 | robwafle | i was under the impression that G711u was the best available to me |
| 04:14.32 | robwafle | ah hah |
| 04:14.47 | bkw_ | but they announced plans to do so |
| 04:15.18 | robwafle | what about 726? |
| 04:15.18 | bkw_ | robwafle: but FreeSWITCH can do G722.1 and G722.1C (32kHz) |
| 04:15.28 | bkw_ | 726-32 Asterisk can do |
| 04:15.32 | bkw_ | which is nice |
| 04:15.35 | robwafle | hey I thought we were in #asterisk :) |
| 04:15.37 | bkw_ | its very good toll quality |
| 04:15.39 | robwafle | j/k |
| 04:15.45 | bkw_ | robwafle: I still use Asteirsk for some stuff too :P |
| 04:15.58 | robwafle | I've not tried FreeSWITCH yet. |
| 04:16.01 | bkw_ | and I have to interface with it all the time |
| 04:16.19 | bkw_ | I currently have a bug open on mantis with Asterisk... wish someone would work with me to finish collecting the traces |
| 04:16.30 | harry_v | I havnt been keeping up with some of the advances in ast but is it taking on new features such as voice recognition capabilities? |
| 04:16.54 | drmessano | That is and should be external |
| 04:17.07 | bkw_ | harry_v: Asterisk does ASR with Lumenvox |
| 04:17.14 | bkw_ | I have been toying with pocketsphinx all day |
| 04:17.21 | bkw_ | updated our mod to use jsgf and a few other things |
| 04:17.24 | harry_v | Lum requires a licence ? |
| 04:17.29 | bkw_ | harry_v: yes |
| 04:17.32 | harry_v | k |
| 04:17.37 | bkw_ | harry_v: what scope of recognition are you looking to do? |
| 04:17.39 | bkw_ | simple things? |
| 04:17.42 | bkw_ | or dictation? |
| 04:17.43 | harry_v | sure |
| 04:18.04 | bkw_ | getting things like numbers and asking simple questions is easy to do with PocketSphinx |
| 04:18.12 | bkw_ | and a guy even wrote a plugin for asterisk for it |
| 04:18.12 | harry_v | dictation would be really interesting |
| 04:18.21 | bkw_ | him and I are working a tiny bit together |
| 04:18.21 | *** join/#asterisk _gm (n=gmustafa@115.186.106.37) |
| 04:18.27 | bkw_ | http://scribblej.com/svn/ |
| 04:18.28 | robwafle | bkw_ would you suggest I download trixswitch? |
| 04:18.28 | bkw_ | thats his |
| 04:18.32 | bkw_ | robwafle: nope |
| 04:18.50 | bkw_ | harry_v: I'm going to help him make his more robust also. |
| 04:19.06 | bkw_ | ScribbleJ and I have been comparing notes |
| 04:19.50 | bkw_ | I'm doing a couple of demos with pocketsphinx to grab flight details by airline, flight number and departure airport |
| 04:19.57 | bkw_ | just gotta get a couple of bugs fixed in pocketsphinx first |
| 04:20.30 | robwafle | so, how should I try freeSWITCH? download the Phoenic tarball and make? |
| 04:21.10 | robwafle | bkw_ have you tried trixbox CE ? |
| 04:21.31 | bkw_ | robwafle: nope |
| 04:21.34 | harry_v | One of the coolest and I mean accurate voice recognition systems I have seen used is by coast mountain buss system here. You verbally tell it where you are and where you want to go and it will tell you which buss will be stopping by and if is late and what time for pickup. |
| 04:21.41 | bkw_ | robwafle: you should go in #freeswitch and I can help you |
| 04:21.45 | robwafle | ok |
| 04:21.52 | bkw_ | harry_v: the one AT&T uses SUCKS |
| 04:21.56 | bkw_ | it never gets jack right |
| 04:22.00 | harry_v | hehe |
| 04:22.09 | harry_v | 1800555 sucks to |
| 04:22.18 | harry_v | sad to see the woman go |
| 04:22.19 | harry_v | ;) |
| 04:22.35 | bkw_ | haha |
| 04:22.43 | harry_v | flirt with a 1800 operator |
| 04:22.44 | harry_v | :) |
| 04:22.45 | bkw_ | who ever wrote the voice app for AT&T DSL support needs to be shot |
| 04:22.47 | dgoner | I usually wind up swearing at it |
| 04:23.08 | bkw_ | most ivr devs put swear menus |
| 04:23.12 | bkw_ | if you swear you get hotlined |
| 04:23.15 | bkw_ | try it on sprint |
| 04:23.20 | bkw_ | curse at the lady |
| 04:23.31 | dgoner | lol I'll keep it in mind next time |
| 04:23.37 | drmessano | "Shits Broke" will get you to Comcast internet support, very quickly |
| 04:23.53 | thehar | dials at&t with a 100 block to spring with tons of voice swearing |
| 04:24.04 | bkw_ | haha |
| 04:24.14 | thehar | s/spring/sprint/ |
| 04:24.27 | bkw_ | queue up the F word |
| 04:24.33 | drmessano | thehar: You should stop.. dont want you to hurt yourself |
| 04:25.05 | thehar | bkw_: heh |
| 04:25.32 | thehar | wow loritab has me not thinking complete sentences |
| 04:25.33 | harry_v | some times ivr or vr does not have its place...especially when your in a hurry. |
| 04:25.45 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 04:25.59 | drmessano | harry_v: Everyone is in a hurry.. always |
| 04:26.06 | dgoner | I wish they would give you an option to choose in the beginning |
| 04:26.13 | harry_v | Some times telemarker calls get past my ivr and was thinking of making one for them ;) |
| 04:26.21 | drmessano | This is hilarious |
| 04:26.33 | drmessano | "If you are not in a hurry and wish to wait, press 5 now" |
| 04:26.36 | harry_v | ohh and it would record there voice |
| 04:26.40 | drmessano | Im sure 5 would get used all the fuckin time |
| 04:27.00 | dgoner | lol |
| 04:27.17 | harry_v | ivr torture |
| 04:27.32 | drmessano | or if they gave an option "If you are in a hurry, press 3" .. do you think someone would NOT press 3? |
| 04:27.40 | drmessano | Think, people |
| 04:27.54 | dgoner | I want the option between ivr and vr |
| 04:28.03 | dgoner | I hate vr |
| 04:29.00 | drmessano | Most systems I use as "Press or say x now" |
| 04:29.09 | drmessano | s/as/have/ |
| 04:29.36 | drmessano | I cant remember one thats purely voice driven, although I have hit one in the past |
| 04:29.50 | drmessano | Just dont remember running into it in ages |
| 04:30.42 | dgoner | I seem to remember the one at Sears being completely VR was annoying |
| 04:30.50 | harry_v | hehe |
| 04:30.58 | harry_v | sell them a ast system |
| 04:32.32 | *** join/#asterisk MaliutaLap (n=biteme@203.171.192.242) |
| 04:49.20 | stabler | I have mentioned this in here before but im still having issues with inbound calls, outbound works fine |
| 04:49.20 | stabler | <PROTECTED> |
| 04:49.20 | stabler | here is a pastebin of my sip.conf extentions.conf and sip debug |
| 04:49.20 | stabler | <PROTECTED> |
| 04:49.20 | stabler | notice in the sip debug how it never registers inbound |
| 04:49.21 | stabler | im not sure if this is normal or not |
| 04:49.23 | stabler | <PROTECTED> |
| 04:49.25 | stabler | let me know if anything else is needed |
| 04:49.27 | stabler | <PROTECTED> |
| 04:49.58 | thehar | what does zaptel.conf say |
| 04:50.05 | ruben23 | hi... |
| 04:50.19 | stabler | thehar: was that a question for me? |
| 04:50.46 | ruben23 | if a isetup an asterisk server with two....ethernet...how do i setup it...that my asterisk server will recognize my wan IP... |
| 04:50.47 | thehar | yup |
| 04:51.18 | stabler | thehar: its a sip trunk |
| 04:51.41 | thehar | oh |
| 04:54.21 | stabler | anyone have any ideas? |
| 04:54.26 | stabler | im completely stumped |
| 04:56.18 | stabler | ruben23: use the "externip=70.xx.x.xxx" command i believe |
| 04:58.03 | *** part/#asterisk ruben23 (n=IT-ADMIN@124.107.3.178) |
| 04:58.19 | stabler | i guess he didnt like my answer :( |
| 05:03.17 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-04874de8342c92ea) |
| 05:04.10 | imcdona | I understand that local channels don't contain 'state' information. When I dial a call using a local channel, the call rings for a short bit then hangs up. Why does it hang up? is there a timer somewhere? |
| 05:05.26 | imcdona | in my situation, I am placing a call to a local channel that in turn calls an external phone. if I don't puck up in time, the call disconnects |
| 05:06.19 | kb3ien | is Dial() timing out? |
| 05:06.44 | imcdona | it is set at 300, at that is not elapsing |
| 05:07.05 | imcdona | Dial("Local/14254669391@from-internal-1f2c,2", "SIP/VP-SIPSJCA/+14254669391|300|") |
| 05:07.15 | imcdona | but when I use a sip phone and not a local channel it works fine |
| 05:07.49 | kb3ien | and when local is called it too has a long timeout. |
| 05:07.55 | *** join/#asterisk outtolunc (n=me@c-71-198-197-201.hsd1.ca.comcast.net) |
| 05:08.22 | imcdona | that is the conclusion I came to as well. But apparently that is not the case |
| 05:08.54 | kb3ien | hoes does from-internal-1f2c call the external phone? |
| 05:09.01 | kb3ien | how. |
| 05:09.59 | imcdona | from internal takes the 425XXXXXXX number throws it into from-internal which then directs the call to a sip provider |
| 05:10.22 | kb3ien | using a new invokation of Dial() ? |
| 05:10.56 | imcdona | yes |
| 05:10.57 | kb3ien | oh i see, your passing the argments to from-internal-1f2c |
| 05:11.18 | imcdona | no that is debug info....internal channel name |
| 05:11.57 | kb3ien | what happen if you put a NoOp in front of and behind the dial and dump the DIALSTATUS ? |
| 05:12.13 | kb3ien | does DIALSTATUS give any clues? |
| 05:12.25 | imcdona | I tried in back of it and saw 'cancel' I will try in front...just a sec |
| 05:13.08 | kb3ien | well its more case of timing there, than the dialstatus. to tell the truth. |
| 05:13.21 | kb3ien | maybe the sip carrier is 'giving up' on routing the call? |
| 05:13.43 | kb3ien | but canceled implies that the calling party gave up. |
| 05:14.01 | imcdona | so becasue when I dial from a SIP channel type (a sip phone) it works |
| 05:14.48 | imcdona | a NoOP is blank before the dial statement when I dumped dialstatus |
| 05:17.26 | kb3ien | hrm. does it work the same if you call answer() before dial() ? |
| 05:18.03 | imcdona | If I answer the phone after the dial before letting it ring for say 7 seconds then it works fine |
| 05:18.18 | imcdona | but less than 300 for sure |
| 05:18.53 | *** join/#asterisk MaliutaLap (n=biteme@203.171.192.189) |
| 05:19.57 | kb3ien | im wondering if calling Answer() will fix the issue. |
| 05:22.53 | imcdona | that won't work in my situation, once the call is indeed answered, it is then passed to anot6her context for processing |
| 05:24.51 | kb3ien | i was wondering if the call was being canceled for you farther up the stack. |
| 05:25.16 | kb3ien | i'm having issues with Dial() not having a no-auto-answer feature myself... |
| 05:25.17 | imcdona | if that what the case, why does it continue to ring? |
| 05:25.42 | kb3ien | no idea. |
| 05:26.54 | *** part/#asterisk dgoner (n=david@mx1.repairpc.net) |
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| 05:38.03 | [TK]D-Fender | kb3ien: Your expectation is illogical... |
| 05:40.13 | *** join/#asterisk Talkradio (i=talkradi@linuxgeneration.ca) |
| 05:40.32 | *** join/#asterisk tengulre (n=tengulre@125.71.208.16) |
| 05:40.40 | kb3ien | which expectaction: i plant an acorn an oak tree grows, i planted a feather and i'm still wating for the hen.... |
| 05:41.29 | *** join/#asterisk GameGamer43 (n=GameGame@cpe-66-67-181-68.rochester.res.rr.com) |
| 05:41.54 | Talkradio | did a basic 2 phone pbx for a client :)) heh weak but a start |
| 05:43.00 | *** part/#asterisk thedonvaughn (i=jvaughn@unaffiliated/printk) |
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| 05:46.39 | kb3ien | are you having fault with my debugging skills or with my desire to make Dial() not answer? |
| 05:50.02 | *** join/#asterisk mw46 (n=marco@64-142-40-224.dsl.static.sonic.net) |
| 05:58.35 | [TK]D-Fender | kb3ien: Make Dial() not answer? What isanity is this? Dial doesn't answer. the device you are CALLING chooses whever otr not to answer. |
| 05:59.12 | [TK]D-Fender | kb3ien: Lets say you call an analog phone on a Zaptel analog channel. I wants to answer all the time. How the hell is * supposed to know that? |
| 05:59.39 | kb3ien | dial() does what answer() does /if/ it hasnt been done already. |
| 05:59.41 | [TK]D-Fender | kb3ien: The deives you are calling does whatever the hell it wants. This YOU can tell them what to do all the time? Don't bet on it |
| 05:59.57 | [TK]D-Fender | kb3ien: Dial doesn't ANSWER calls, it PLACES THEM. |
| 06:00.03 | [TK]D-Fender | kb3ien: You are completely ass-backwards |
| 06:00.39 | kb3ien | if the Dial() command is used in the context of processing an inbound call, the inbound call is Answer()ed when the outbound call is. |
| 06:00.59 | kb3ien | that's not what i want. i want to Answer the inbound call when i call answer() only. |
| 06:01.01 | [TK]D-Fender | kb3ien: Not unless you did something to cause this. |
| 06:01.24 | [TK]D-Fender | kb3ien: For that you'd have to actually lok at what you're doing to find out why it answered. |
| 06:01.49 | [TK]D-Fender | kb3ien: I'm sure its quite blatant to instant you pastebin the output of your call |
| 06:03.53 | mw46 | is there any news on `external user authentication' for sip/iax. be it ldap, radius or something else? |
| 06:04.21 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 06:04.58 | [TK]D-Fender | mw46: You can use LDAP for SIP peers, etc |
| 06:05.02 | mw46 | I found http://bugs.digium.com/view.php?id=5424 but it looks like there was not much going on lately:-( |
| 06:05.50 | [TK]D-Fender | mw46: Go read the book's chapters on reealtime... |
| 06:05.52 | [TK]D-Fender | ~book |
| 06:05.52 | jbot | somebody said book was probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 06:06.06 | [TK]D-Fender | kb3ien: So, any odds of seeing this broken call? |
| 06:06.58 | kb3ien | working on it.... |
| 06:07.07 | k-man | so i have set up asterisk, and I can call out from my sip phone, but incoming calls does not work. any idea how to track down where its failing? i should add that the asterisk server is behind NAT |
| 06:07.34 | [TK]D-Fender | k-man: with no info on what you've really looked for? could be ANYTHING. |
| 06:08.03 | [TK]D-Fender | k-man: just jsut start by figuring its as bad as can be and you have no clue how to set up * to work from bnehind NAT. |
| 06:08.05 | [TK]D-Fender | ~sipnat |
| 06:08.06 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 06:08.07 | [TK]D-Fender | ^^^ |
| 06:08.10 | [TK]D-Fender | k-man: Go read. |
| 06:08.45 | k-man | thanks [TK]D-Fender |
| 06:09.28 | *** join/#asterisk Subdolus (n=subby@subby.afraid.org) |
| 06:11.29 | mw46 | [TK]D-Fender: but that would mean clear text passwds in the ldap:-( |
| 06:12.14 | [TK]D-Fender | mw46: Where does it say that? |
| 06:12.31 | mw46 | I've actually a little different problem. I would need a real `external program' to validate the username/`password' because I would like to use a web `session cookie' ;-) |
| 06:12.54 | [TK]D-Fender | mw46: You've got the source, just like the rest of us. Best of luck |
| 06:13.07 | mw46 | http://www-rocq.inria.fr/who/Philippe.Sultan/Asterisk/asterisk_sip_external_authentication.html |
| 06:14.19 | mw46 | that's not exactly the answer I would have liked, but I don't fear that either;-) |
| 06:17.46 | kb3ien | phone calls... grr.. |
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| 06:22.28 | kb3ien | [TK]D-Fender : http://www.voip-info.org/wiki-Asterisk+cmd+Dial says "The first channel that answers "wins", and all the other outgoing channels are hung up. The originating channel that triggered this Dial command is then Answered, if necessary, and the two channels are connected together ("bridged") allowing a conversation to take place between them." is this untrue? |
| 06:22.41 | kb3ien | i ask b/c redacting the logs it taking a while... |
| 06:23.55 | [TK]D-Fender | kb3ien: You are grasping at straws, all of them wrong. |
| 06:24.22 | [TK]D-Fender | kb3ien: I know full well how Dial works, so forget any excuses until you show a precise call. |
| 06:24.46 | kb3ien | so that is wrong too? |
| 06:25.08 | [TK]D-Fender | kb3ien: "The first channel that answers "wins" <- this applies to one of MULTIPLE channels you are CALLING. That the CALLER'S channel |
| 06:26.16 | [TK]D-Fender | kb3ien: Dial(SIP/a&SIP/b). If "a" answers, "b" gets cut off. This has nothing to do with the CALLER, but how one of multiple callee's reacts |
| 06:26.45 | carrar | What if A forwards to the calling extension!! ;) |
| 06:27.16 | [TK]D-Fender | carrar: But Egon.... I though you said crossing the streams was baaaaaaddddd |
| 06:27.36 | carrar | Asterisk Vortex! |
| 06:27.42 | kb3ien | okay. i'm getting the gist of what it means, and checking for any stray Answers() happeing before their time. |
| 06:28.00 | [TK]D-Fender | kb3ien: MANY other things than that. |
| 06:28.29 | carrar | kb3ien, it's really fun when 'a' is a cell and the cell's voicemail picks up |
| 06:28.50 | carrar | or a is forwarding |
| 06:28.54 | carrar | to crapland |
| 06:29.42 | carrar | "i thought I would just forward my phone to the receptionist" |
| 06:29.55 | carrar | or to the ring all extension |
| 06:30.26 | carrar | goes bcak to his hole |
| 06:30.56 | [TK]D-Fender | pours concrete over top and rivits a person-hole cover to cap it |
| 06:31.38 | carrar | escapes out the other side... someplace in Japan |
| 06:31.44 | thehar | carrar: ! :) |
| 06:31.52 | thehar | did you happy to get that purty debug agi written ever? |
| 06:31.59 | carrar | I started |
| 06:32.02 | thehar | yay |
| 06:32.11 | carrar | but ran into some issues with the way I make calls |
| 06:32.15 | thehar | ah |
| 06:32.24 | carrar | and I don't like RED letters on my console |
| 06:32.29 | thehar | me either |
| 06:33.02 | carrar | but you can have what I did, never did the db part, but thats easy once you have the vars |
| 06:33.15 | carrar | all the vars work |
| 06:33.25 | carrar | too many of them actually so you can trim it down |
| 06:33.36 | thehar | sweet. i wouldn't mind it |
| 06:34.40 | stabler | [TK]D-Fender: i need your help |
| 06:34.42 | stabler | I have mentioned this in here before but im still having issues with inbound calls, outbound works fine |
| 06:34.42 | stabler | <PROTECTED> |
| 06:34.42 | stabler | here is a pastebin of my sip.conf extentions.conf and sip debug |
| 06:34.42 | stabler | <PROTECTED> |
| 06:34.42 | stabler | notice in the sip debug how it never registers inbound |
| 06:34.44 | stabler | im not sure if this is normal or not |
| 06:34.46 | stabler | <PROTECTED> |
| 06:34.48 | stabler | let me know if anything else is needed |
| 06:34.50 | stabler | <PROTECTED> |
| 06:34.54 | [TK]D-Fender | stabler: don't SPAM IN HERE |
| 06:35.18 | stabler | thats not spam |
| 06:35.22 | carrar | http://www.osburn.com/hangupstats |
| 06:35.24 | carrar | there you go |
| 06:35.33 | thehar | thanks man |
| 06:35.55 | thehar | jebus vars |
| 06:36.01 | carrar | a lot of those vars will never get populated so you an remove them |
| 06:36.42 | thehar | i should try it out now.. since the pbx is slow |
| 06:37.12 | stabler | its a question |
| 06:37.20 | [TK]D-Fender | stabler: bindport=5060,5070 <- you can't specify multiple ports last I checked |
| 06:37.35 | carrar | reload it |
| 06:37.44 | [TK]D-Fender | stabler: and I don't see a call attempt in there |
| 06:38.01 | carrar | thehar, right now it just spits everything out to /tmp/printoutagivars |
| 06:38.11 | thehar | just saw that |
| 06:38.17 | stabler | [TK]D-Fender: thats because an inbound call doesnt trigger anything on the sip debug |
| 06:38.47 | carrar | pack what you need and instert in your DB as a associative array |
| 06:38.57 | carrar | which was where I was going with that |
| 06:39.28 | stabler | [TK]D-Fender: i can get output for an outbound call if you want it |
| 06:39.44 | thehar | dur this pbx i was going ot use doesn't have agi.pm =) |
| 06:39.44 | carrar | thehar, also need to parse out $chaninfo{'rtpqos'} |
| 06:39.50 | carrar | but thats easy |
| 06:39.52 | thehar | mmhm |
| 06:39.56 | stabler | [TK]D-Fender: sip show registry |
| 06:40.04 | carrar | You will see |
| 06:40.04 | stabler | [TK]D-Fender: shows its registered |
| 06:40.40 | thehar | either rate.. thanks |
| 06:40.43 | thehar | appreciate it |
| 06:40.54 | thehar | at least you started.. i wouldn't have gotten that far |
| 06:41.01 | *** join/#asterisk Maliuta (n=scooby@kiev.lusan.id.au) |
| 06:41.28 | kb3ien | okay here we go! |
| 06:41.47 | stabler | [TK]D-Fender: what port should i bind since my sip provider requires 5070? |
| 06:42.25 | [TK]D-Fender | stabler: I really don't ahve any motivation to deal with MagicJack issues. |
| 06:42.50 | stabler | [TK]D-Fender: ok.. fair enough... thanks anyway |
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| 06:43.39 | stabler | [TK]D-Fender: i was simply wanting to use it to further my education/knowledge... but im stuck |
| 06:44.46 | thehar | i think my pinot went bad :( |
| 06:44.52 | thehar | this glass tastes like poop |
| 06:45.11 | thehar | stupid failing wine fridge! |
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| 06:46.06 | carrar | I just had wine |
| 06:46.32 | carrar | I write great when I've had alot then the next day can't figure out what i wrote |
| 06:46.45 | thehar | ha |
| 06:47.16 | stabler | [TK]D-Fender: can you atleast confirm if its possible or not to get inbound to work with MJ on asterisk? I have a feeling it may not be possible... |
| 06:47.45 | stabler | [TK]D-Fender: just knowing that would mean alot to me |
| 06:50.56 | kb3ien | my ${ARGx} are getting lost/corupted too. i'll be a while on this one |
| 06:51.47 | [TK]D-Fender | stabler: Havven't you found a dozen blogs with isttructions on this already? |
| 06:51.53 | [TK]D-Fender | instructions* |
| 06:52.43 | stabler | [TK]D-Fender: yes, ive followed them and re-followed them.. which got my outbound working perfectly |
| 06:53.33 | [TK]D-Fender | stabler: Keep reading then. |
| 06:53.46 | [TK]D-Fender | kb3ien: Got something yet? |
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| 06:54.18 | stabler | [TK]D-Fender: i shall |
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| 07:04.36 | [TK]D-Fender | ok, bed time, later all |
| 07:09.02 | kb3ien | fyi |
| 07:09.16 | kb3ien | i figured out where my code was going awry. |
| 07:09.53 | kb3ien | one nice typo reinforced my a misreading of some docs lead me down a wrong path. I'll tell him tomorrow. |
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| 07:28.49 | carrar | thehar |
| 07:28.50 | trijezdci | anyone here using BRIstuff? |
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| 07:55.50 | trijezdci | does anybody know what the zaptel message "no version for "zt_receive" found: kernel tainted." is about? It seems to be everywhere, no matter what hardware folks use and nobody ever seems to care about this but I am curious why this comes up all the time |
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| 08:16.23 | steerpike | hi, if you have firewall issues, can you use just about any stun server? |
| 08:16.47 | steerpike | or does the sip provider have to run the stun servere? |
| 08:18.24 | kaldemar | you don't even need stun if you control the firewall. |
| 08:18.27 | kaldemar | ~sipnat |
| 08:18.28 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
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| 08:29.07 | jad_jay | Hi everybody |
| 08:29.30 | jad_jay | Is there any possibility to connect a cellphone to asterisk with usb cable? |
| 08:31.40 | stabler | jad_jay: you can via bluetooth |
| 08:32.02 | stabler | and use it as an inbound/outbound trunk |
| 08:32.07 | jad_jay | Ok so it's the only possibility |
| 08:32.12 | jad_jay | cool |
| 08:32.34 | stabler | jad_jay: i dont know of any others |
| 08:32.39 | stabler | jad_jay: maybe someone else does |
| 08:35.13 | mcnobody | jad_jay: GSM gateways are an option also |
| 08:35.23 | *** join/#asterisk scruz (n=scruz@41.220.73.170) |
| 08:35.35 | scruz | morning |
| 08:35.56 | mcnobody | jad_jay: For example 2N Voiceblue. Connects to Asterisk with SIP @ ethernet. |
| 08:40.15 | *** join/#asterisk trijezdci (n=trijezdc@f8a01-0097.din.or.jp) |
| 08:40.56 | yang | Yes, you might want to take a look at a cheaper GSM gateway solution , like portech |
| 08:41.24 | yang | 2N are really great products but in the more expenssive range |
| 08:41.49 | *** join/#asterisk botox93 (n=botox93@213.221.82.242) |
| 08:42.08 | mcnobody | yang: Have you used Protech GSM gateways? |
| 08:42.29 | mcnobody | yang: our customer has used VoiceBlue a couple of years... |
| 08:42.34 | *** join/#asterisk frk2 (n=frk2@zivios/member/fkhan) |
| 08:43.06 | yang | yes, we use them....some of them also broke unexpectedly, but its a cheap solution for a home-voip, not ment for any professional use |
| 08:43.43 | yang | I think they cost around 150 USD for a one-port node |
| 08:44.05 | mcnobody | yang: Not bad. |
| 08:44.37 | mcnobody | yang: Protech seems to have alse PRI GSM channel bank. |
| 08:45.09 | yang | yes, they make various products |
| 08:45.33 | yang | ~portech |
| 08:45.52 | mcnobody | yang: It's sure for professional use... But have no idea of quality.. |
| 08:46.25 | yang | Well, you cannot really compare them with 2N solutions neither by quality or the price |
| 08:46.50 | mcnobody | yang: Ok. |
| 08:47.49 | mcnobody | yang: Our customer has 4 channel VoiceBlue and they have been very happy for it. Some little stranges there are with older Asterisk.. |
| 08:48.12 | jad_jay | mcnobody: yes but i first look about unexpensive solution :) The 2N Voiceblue is about 1000$ isn't it? |
| 08:48.13 | scruz | i'm having trouble building dahdi. here's the output after i run make: http://pastebin.com/dd65f5dd |
| 08:49.06 | yang | jad_jay: www.portech.com.tw |
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| 08:49.50 | mcnobody | jad_jay: Actually I have no idea of the price of VoiceBlue =) Our customer has leased it from somewhere else. We just configured it as a part of the pbx. |
| 08:50.33 | mcnobody | jad_jay: But I know that 4 channel VoiceBlue have been cost effect solution for gsm calls there. |
| 08:50.45 | jad_jay | yang: And i just need a card with an account |
| 08:51.31 | jad_jay | yang & mcnobody : thanks but i need prices |
| 08:54.20 | yang | jad_jay: take a look at their distributor websites |
| 08:55.41 | steerpike | my sip provider's port is 5070, my client port is 5060, on the router i just port forward 5060, and set the QoS priority for 5070 |
| 08:58.48 | *** join/#asterisk shtoom (n=shtoom@121.246.167.147) |
| 09:00.32 | kaldemar | steerpike: you need to forward rtp ports too (by default 10000-20000 in asterisk) for inbound audio to work. |
| 09:00.50 | steerpike | oh yeah, forgot about that one |
| 09:03.52 | steerpike | so QoS only applies to a remote server's port? |
| 09:06.12 | kaldemar | depends on what direction it is configured, but trying to use QoS over the internet is pretty much useless. you can only control your own router on the whole path. |
| 09:06.37 | steerpike | yeah, i've noticed it doesn't make a whole lot of difference |
| 09:11.38 | *** part/#asterisk lanning (n=lanning@173.8.187.197) |
| 09:11.46 | jad_jay | yang & mcnobody : Okay for a one way GSM it is about 200€ (and this is C°°L :) |
| 09:11.52 | awk | hi, anyone have a path where I can download fpm-calm-river.wav fpm-sunshine.wav fpm-world-mix.wav in mp3 format, yes I can convert, but that is not what i'm asking. |
| 09:12.12 | jad_jay | Porttech MV370 |
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| 09:21.36 | kaldemar | awk: http://downloads.digium.com/pub/telephony/sounds/ |
| 09:23.53 | awk | kaldemar thanks |
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| 09:35.10 | *** join/#asterisk stewbaby (n=stewart@ip-217-204-65-78.easynet.co.uk) |
| 09:36.03 | jad_jay | Where can i find anything about attended call transfert |
| 09:36.19 | jad_jay | this is for you jbot :) |
| 09:40.31 | jad_jay | No one can help me on attended call transfert ?????? O_o |
| 09:41.23 | kaldemar | maybe if you ask a real question |
| 09:41.48 | scruz | jad_jay: try google |
| 09:42.48 | kaldemar | otherwise it will just be http://www.google.com/search?q=asterisk+attended+transfer |
| 09:50.36 | *** join/#asterisk kaldemar (n=kalde@vipunen.hut.fi) |
| 09:51.38 | grEvenX | hm |
| 09:52.08 | grEvenX | peaking on 60-70% CPU utilization with only 18 active channels |
| 09:52.16 | grEvenX | that doesn't sound good, does it? |
| 09:52.23 | *** join/#asterisk lbt (n=david@78.32.229.233) |
| 09:53.30 | scruz | grevenx: what codecs are in use? if there's a lot of transcoding goind on, it might be 'normal' |
| 09:53.37 | scruz | *going |
| 09:53.53 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 09:54.18 | scruz | ignores the refusal of dahdi to build since it seems nobody can help him with boulding it |
| 09:54.26 | scruz | *building |
| 09:54.41 | grEvenX | scruz: uLaw |
| 09:54.41 | *** join/#asterisk jmkgreen (n=chatzill@fentech.gotadsl.co.uk) |
| 09:55.12 | grEvenX | scruz: ulaw|alaw|h261|h263|h263p |
| 09:55.33 | grEvenX | there's no video going through, so it's either a/u-law |
| 10:06.34 | *** join/#asterisk Hobadee (n=eric@c-98-234-44-132.hsd1.ca.comcast.net) |
| 10:07.38 | scruz | grEvenX: well, i'm more-or-less an asterisk newbie. i only work with it when my boss requires me to. and since he shifts my responsibilities quite often, i get moved around quite often |
| 10:12.58 | Hobadee | so I have a potentially stupid question here, but I've been googling for hours and can't figure it out |
| 10:13.22 | *** join/#asterisk Czindy (n=Czindy@91.120.30.42) |
| 10:13.46 | Hobadee | I just downloaded/installed AsterixNOW 1.5 Beta - installed fine and everything, but can't login to the webgui |
| 10:14.15 | Hobadee | I've tried countless username/password combinations - nothing seems to work... |
| 10:22.04 | Czindy | Hello, |
| 10:22.42 | Czindy | I have some problem with using realtime extensions. Could somebody help on this please? |
| 10:23.27 | jmkgreen | We have a large number of the following in our logs from last night: [Feb 19 19:48:25] WARNING[17686] chan_iax2.c: Max retries exceeded to host 217.14.138.130 on IAX2/trunk_2-12352 (type = 6, subclass = 11, ts=40015, seqno=9) |
| 10:23.43 | jmkgreen | Does anyone off hand know where we should be headed to fix this? |
| 10:24.11 | jmkgreen | I see a bug in Mantis which may be the same issue: http://bugs.digium.com/view.php?id=14008&nbn=1 |
| 10:25.19 | jmkgreen | That's purely firing off 30 prerecorded voice calls to PSTN lines pretty much simultaneously through two different providers. It is working fine with two calls. |
| 10:25.45 | *** join/#asterisk Psychobilly (n=moi@adsl65-119.kln.forthnet.gr) |
| 10:41.34 | jermey_g | guys, this bridge app has really let me down |
| 10:41.41 | grEvenX | hm |
| 10:41.58 | grEvenX | I have a script polling the AJAM from HTTP in 2 sec intervalls |
| 10:42.38 | jermey_g | i bridge an incoming call on an extension to an on going sip call which has its one party loosing audio. That phone just can't decode the rtp from bridge |
| 10:42.49 | jermey_g | anyone like to see a wireshark trace on this and my config |
| 10:43.19 | grEvenX | "manager show connected" lists a shitloads of connections |
| 10:43.40 | grEvenX | am I doing something wrong with the way I connect to the AJAM? |
| 10:43.47 | grEvenX | I get like 2000 manager connections over time |
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| 10:46.10 | *** join/#asterisk flohack (n=fhackenb@lancelot.acoveo.com) |
| 10:47.05 | flohack | Is there a flag which has to be set to get debugging information in res_agi.so? Debug info is present in app_stack.so for example but not res_agi.so. I recompiled all object and shared object files in res/. |
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| 11:23.19 | jermey_g | i couldn't find anything about Allow promsic. redir in the the book. Can anyone explain what it is? |
| 11:26.29 | *** join/#asterisk BBHoss (n=bbhoss@68.159.148.173) |
| 11:27.25 | jermey_g | sorry, i found it. |
| 11:29.11 | zamba | i'm getting the following error: "chan_sip.c:13865 handle_request_invite: Call from '' to extension 'test' rejected because extension not found." |
| 11:29.17 | zamba | how can i figure out what's wrong here? |
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| 11:36.09 | jermey_g | zamba:show your extensions.conf |
| 11:36.17 | jermey_g | pastebin |
| 11:40.36 | flohack | I'm using an AGI script to forward calls to a queue (using Dial with a local channel). As soon as a timeout on the queue occurs, I need to do some processing and put the caller into the same queue again. For reporting purposes, I need to detect the caller entering the same queue again. Because of the Dial, however, the unique id is different everytime the caller enters the queue. |
| 11:41.15 | flohack | Is there a way to enter the dialplan from within an agi script without spawning a new channel? |
| 11:44.31 | jermey_g | what happens if i dial the parking extension from one of phones :) |
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| 11:55.29 | flohack | is away: Lunch |
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| 11:58.54 | grEvenX | anyone using the AJAM? |
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| 12:25.03 | plundra | What's the easiest way of printing variables during a call? Preferably so the output lands in the asterisk console ('asterisk -r') |
| 12:25.53 | plundra | Verbose()! Thanks :-) |
| 12:26.59 | *** join/#asterisk arnuld (n=arnuld@unaffiliated/arnuld) |
| 12:27.09 | arnuld | My compnay is using Asterisk |
| 12:27.43 | arnuld | ... and it is my job to understand the asterisk API, configure it properly, write call files etc etc |
| 12:27.53 | arnuld | I can't find the documentation of aserisk |
| 12:27.55 | arnuld | oon net |
| 12:28.00 | arnuld | not even on asterisk.org |
| 12:28.02 | arnuld | I mean |
| 12:28.10 | arnuld | NO Reference Manual or something |
| 12:28.21 | arnuld | anythign for newbie ? |
| 12:28.29 | arnuld | (I am basically a programmer) |
| 12:30.21 | mcnobody | arnuld: http://www.voip-info.org/ |
| 12:30.30 | arnuld | I knew that |
| 12:30.37 | jermey_g | ~thebook |
| 12:30.38 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 12:30.39 | arnuld | but I did not know if it was trustable place |
| 12:31.06 | jermey_g | arnuld:its in the * tar ball, ever did a make progdocs |
| 12:31.13 | mcnobody | arnuld: It's quite trustable. Most big problem there is different Asterisk versions. |
| 12:31.14 | arnuld | yep |
| 12:31.20 | arnuld | but they are like soingle page docs |
| 12:31.28 | jermey_g | arnuld:and rest of it is in people's minds at #asterisk-dev |
| 12:31.28 | arnuld | mcnobody: ok |
| 12:31.47 | mcnobody | But I have to go now.. |
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| 12:49.22 | plundra | Hmm, is there something undoable with having three registers and/or three sip peers with the same remote host? |
| 12:50.04 | plundra | Because incoming calls on the differente lines are fine, but, I can't call out using any other then the first "line". |
| 12:51.28 | flohack | is back |
| 12:53.12 | flohack | plundra: I'm using three sip phones (pjsip based) on the same host without any problems |
| 12:53.39 | flohack | Have a look at the asterisk cli with 'sudo asterisk -rvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvv' |
| 12:54.01 | *** part/#asterisk lenne_dk (n=Leif@cpe.atm2-0-74391.hknxx4.customer.tele.dk) |
| 12:54.02 | flohack | And see if asterisk complains when making the outgoing call |
| 12:54.07 | plundra | flohack: Yeah that's fine I guess, but I mean, I assume it uses the same port/whatever for everything, or something. I dunno. :-) |
| 12:54.36 | plundra | I've tried using sip debug, but I didn't really make any sense in it. I want coffee too... |
| 12:54.55 | flohack | Which client are you using? |
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| 12:56.50 | plundra | I don't really see why the client makes any different? :) It's the way out that gets borked. I'm using both a zoiper softphone (via iax2) and a 7940 (via sip) |
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| 12:57.58 | flohack | describe your setup in detail, please |
| 12:58.15 | flohack | Is there a way to enter the dialplan from within an agi script without spawning a new channel? |
| 13:01.30 | plundra | flohack: Ok, I've got three numbers from a provider, coming in the form av three different username/passwords, but the same host. In sip.conf I have a register for each account, aswell as a [numN], type=peer, <account info> |
| 13:02.47 | plundra | Up until now, it's been working great, calling in on any of them, each with a different extension etc, works great. Also, calling out (Dial(SIP/${EXTEN:1}@num1);) has been working great, using num1 that is. |
| 13:02.50 | flohack | ok and when saying 'I can't call out using any other then the first "line"' what does that mean? which line? |
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| 13:03.24 | plundra | The one with the lowest actual phonenumber, which also has been in order in sip.conf |
| 13:03.25 | flohack | do you mean establishing one call via sip using @num1 works, but the second call fails? |
| 13:03.42 | plundra | No, when I want to call via @num2 |
| 13:04.19 | flohack | you establish your first call with: 'Dial(SIP/${EXTEN:1}@num1' and the second with 'Dial(SIP/${EXTEN:1}@num2' and the second call fails? |
| 13:04.43 | plundra | I'm just changing the dialplan to use num2 always. |
| 13:04.54 | plundra | I havn't managed to get a single call out using that number/account. |
| 13:05.55 | flohack | your best bet it probably to look at the sip protocol if the asterisk console does not provide enough information. Wireshark is REALLY nice for that. It can even display a graphical protocol session. |
| 13:06.24 | plundra | Yeah, might have to do that. |
| 13:06.36 | flohack | check which response (404 for example) comes back from your sip provider right after the INVITE |
| 13:06.48 | plundra | Also, I'll stop asterisk completly and wait for all registrations or whatever in all ends to timeouot. |
| 13:07.41 | plundra | And then try calling out using num2 or num3, completly leaving num1 out of the configuration. And then debug and see if I get any sip-traffic calling to num1, amongst many other things :-P |
| 13:07.59 | flohack | good idea |
| 13:08.12 | plundra | B |
| 13:08.42 | plundra | Oops. Well, thanks so far :-) Now I'll get some coffee and wait out he timeouts :-P (That's my official excuse ;)) |
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| 13:10.19 | arekm | \ |
| 13:11.02 | arekm | hi, I have some weird sip phones (8level ipp822) that have very loud call waiting indicator. I guess that's phone property but maybe it's not and something can be done on asterisk level? |
| 13:11.18 | arekm | The conversation is "broken" due to this indicator |
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| 13:14.27 | gambler1 | Hello, I'am having a 484 (address incomplete) problem when dialing carrier. Does anyone here have a hint for this? I have been looking and it seams that problem is related to * dial application |
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| 13:48.22 | jad_jay | What is this error about ? [Feb 20 14:48:01] WARNING[26819]: app_meetme.c:800 build_conf: Unable to open pseudo devic |
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| 13:54.58 | [TK]D-Fender | jad_jay: Looks like you don't have dahdi installed and configured for a timing source |
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| 13:57.00 | jmkgreen | Has anyone got any idea what might cause this: [Feb 19 19:48:25] WARNING[17686] chan_iax2.c: Max retries exceeded to host 217.14.138.130 on IAX2/trunk_2-12352 (type = 6, subclass = 11, ts=40015, seqno=9) |
| 13:57.24 | jmkgreen | seems to prevent audio in answered calls |
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| 13:59.52 | eric2 | all my calls come through a TDM card on box A and are then fired off to box B, when the network connection is down on box A, I need * to reject the call as the telco will fire off the call to box B over SIP |
| 14:00.20 | eric2 | what's the best way for * on box A to reject the call when the network is down so that the call goes directly from telco to box B? |
| 14:01.24 | [TK]D-Fender | eric2: Don't answer the call when it comes in. Dial B. On failure call Conjestion() or something |
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| 14:01.40 | [TK]D-Fender | eric2: Now this depends on what TECH is being used and how the telco reacts. |
| 14:02.30 | eric2 | ok, I"ll inquire with the telco on the best way to signal back to them - the rejected call |
| 14:02.41 | flohack | Is there a way to enter the dialplan from within an agi script without spawning a new channel? |
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| 14:14.06 | [TK]D-Fender | flohack: go read the chapter on AGI, three are clear commands for specifying where to resume dialplan. |
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| 14:21.01 | flohack | [TK]D-Fender: I know how to continue the dialplan after terminating the script, but I need to stay within the script and need to jump to the dialplan, regaining control as soon as the dialplan finished. |
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| 14:24.35 | Katty | hai |
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| 14:27.39 | fiddur | Hi. We have a inbound call center customer who looks more at service level than on abandoned calls or speed of answer... Is there an easy way to have asterisk let callers in a queue who haven't yet reached the service level time get through to a free queue member earlier, overriding the FIFO principle? (without modifying call prio constantly from a manager...) |
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| 14:38.57 | deeperror | morning, I just updated from 1.4.21.1 -> 1.4.23 and now it seems that dtmf isn't being heard on the receiving party such as when dialing into an ivr and entering extensions. Do I need to modify something? When typing 1234567890 the other side only sees 13579 |
| 14:38.58 | cjk | hi, i seem to have a problem where the fax machine connected to an fxs port sends the number but it arrives differently (few digits are wrong) in asterisk, has anyone seen this? |
| 14:40.56 | Katty | so quiet this morning |
| 14:43.17 | plundra | flohack: Mkay, so here's what I got now, when I try to use the "other" (num2/num3), I get a 403 Forbidden after it tries the Proxy_Authentication, and it gives up. |
| 14:43.32 | [TK]D-Fender | flohack: You'd have to dial a local channel, but I doubt that'll work for your needs. AGI does not sit behind dialplan, it runs instead of it. |
| 14:43.57 | deeperror | What changes have been made regarding dtmf after upgrading from 1.4.21 -> 1.4.23? |
| 14:44.05 | plundra | flohack: So I figured maybe the password was wrong, so I tried removeing it in the registry-line and complained about it directly, so the password is indeed correct. |
| 14:45.20 | plundra | flohack: But still, I can't dial out using it. The non-verbose output is here; http://pastie.org/395093.txt?key=ifcjxmnja7ob66irjklaya |
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| 14:49.14 | plundra | I tried using the account directly in another sip-client, and it works fine to dial out. So what I get is, that only one of the accounts works for dialing out from asterisk, and they are configured in the exact same way. And the account obviously works, since I can dial out from sip-client directly, and register with it in asterisk (and dial in) |
| 14:49.29 | eric2 | [TK]D-Fender what's the quick and dirty way to reject all calls over TDM so the SIP backup will take over? |
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| 14:51.56 | dvdevel | eric2, how about 'module unload chan_dahdi.so' |
| 14:52.32 | Kobaz | eric2: you can send back BUSY or CONGESTION, or something like that |
| 14:52.58 | Kobaz | or yeah, unload the module so it doesn't get used at all |
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| 14:54.10 | Kobaz | might as well take the card out too |
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| 14:56.12 | angryuser | hello what kind of sip error shold be used when the number is bad ? busy ? and when network is unreachable ? |
| 14:56.30 | angryuser | or let's say the number is not assigned |
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| 14:56.35 | angryuser | 404 ? |
| 14:56.50 | Kobaz | i would say 404 |
| 14:57.22 | deeperror | Having an issue after upgrading from 1.4.21 -> 1.4.23 with dtmf not being picked up on the called IVR. Any clues? www.pastebin.ca/1342908 |
| 14:58.04 | [TK]D-Fender | eric2: I asked you to be specific about your connection |
| 14:58.39 | angryuser | Kobaz: i can reach the destination network but the number is not assigned (mobile operator) i got the standart 3 bips and the "number is not assigned" but only when i call with BRI, mo VOIP provider send me 403 forbidden in place |
| 14:59.00 | angryuser | should i blame the voip provider ? |
| 14:59.30 | Katty | [TK]D-Fender: mew. |
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| 15:00.01 | [TK]D-Fender | Katty: Mew. |
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| 15:05.04 | deeperror | Having an issue with dtmf not being picked up on the called IVR. www.pastebin.ca/1342908 In sip.conf i've got dtmfmode=rfc2833 |
| 15:06.23 | Kobaz | angryuser: mmm, 403 would be the wrong response i would think, that's more of like "this number exists but you can't use it" |
| 15:07.04 | Katty | hello, how to use asterisk??? |
| 15:07.20 | Kobaz | licks Katty |
| 15:07.41 | angryuser | Kobaz: so it's more like 404, i think |
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| 15:08.11 | Katty | where's jaytee |
| 15:08.46 | Kobaz | i would think it should be a 404 |
| 15:10.18 | angryuser | Katty: very simple, you click Options>Use Asterisk :) |
| 15:12.17 | jaytee | Katty, I'm over there ------> |
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| 15:14.01 | Katty | angryuser: huH? |
| 15:14.05 | Katty | hugs jaytee |
| 15:14.13 | jaytee | hugs Katty |
| 15:14.50 | riddlebox | hey all |
| 15:17.49 | Katty | hey riddle |
| 15:17.58 | Katty | i found an article about agent orange on reddit today. it's sad. |
| 15:18.35 | Katty | i'd like to decapitate who ever authorized using it |
| 15:18.44 | Katty | wait, no |
| 15:18.47 | Katty | let's put agent orange ON them |
| 15:18.53 | *** join/#asterisk KavanS (i=proxy@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 15:18.58 | Katty | and let them suffer miserably for the rest of their life. |
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| 15:21.53 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:22.13 | Katty | hi mog |
| 15:23.16 | *** join/#asterisk LuisTorres (n=chatzill@81.84.224.208) |
| 15:23.23 | LuisTorres | Hi all |
| 15:23.32 | mog | hi Katty |
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| 15:24.59 | LuisTorres | anyone has sucessfuly installed cepstral or swift? |
| 15:25.09 | FunkyGMT | LuisTorres yes. |
| 15:25.10 | madsara | Hey, quick question re: DTMF. Inbound SIP having an issue where DTMF isn't being detected in a Read() call. In sip.conf I've disabled all, enabled ulaw, and set dtmfmode=rfc2833 (Actually I've tried all possible settings for dtmfmode). Any suggestions as to something I may be missing? |
| 15:25.50 | LuisTorres | can you help me with a few questions? |
| 15:26.01 | FunkyGMT | Sure |
| 15:26.57 | LuisTorres | first I got the app_swift.c and compile it |
| 15:27.15 | LuisTorres | copied to modules dir |
| 15:27.46 | jaytee | I currently have 141 phones on my primary * server and by the end of the day I'll have 147. |
| 15:28.52 | FunkyGMT | jaytee: Nice! |
| 15:29.01 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 15:30.12 | jaytee | FunkyGMT, yep and by next thursday or friday I hit the magic number of 173 and I pull the plug on the Nortel Option 11C |
| 15:31.54 | FunkyGMT | rockon! |
| 15:33.17 | *** join/#asterisk ming_zym (n=ming_zym@210.192.102.198) |
| 15:34.35 | Talkradio | 173 extensions? |
| 15:34.45 | Talkradio | nice |
| 15:35.08 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) |
| 15:35.17 | Talkradio | what kind of specs does it take to support that many users? |
| 15:36.52 | [TK]D-Fender | Talkradio: a basic PC I can pick up for $300 can do this. |
| 15:37.11 | [TK]D-Fender | Talkradio: Everything depends on what they are doing. just tossing calls around from a LAN to say a PRI is nothing |
| 15:39.25 | Talkradio | so what kind of specs would you throw at such a setup [TK]D-Fender |
| 15:40.57 | [TK]D-Fender | Talkradio: Depends on the "other" factors. Nominal in mind, basic C2D w/ RAID1, 4 BG ram (just because tis so cheap. |
| 15:41.06 | [TK]D-Fender | Talkradio: Ok, maybe 400$ or so. |
| 15:41.51 | Talkradio | so a nice little dell 330 / 360 |
| 15:42.11 | Katty | or build one yourself off newegg |
| 15:42.32 | Talkradio | so far all my installs are on dell 330's just because they are cheap and it's what the call center edition i first installed from pbxtra was |
| 15:42.47 | Katty | well keeping them all the same is good for part swapping |
| 15:42.59 | Katty | i don't like dealing with dell tho |
| 15:43.08 | Talkradio | i just went with sagnoma 101D's and a200's and dells and i have my fingers crossed heh |
| 15:43.19 | Katty | i like sangoma |
| 15:43.20 | Talkradio | Katty yeah dealing with them can suck |
| 15:43.34 | *** join/#asterisk yang (i=yang@CAcert/Assurer/pdpc.supporter.base.yang) |
| 15:43.40 | Katty | dealing with sangoma people is a breeeeeze |
| 15:43.50 | Talkradio | i meant dell |
| 15:43.56 | Katty | ya i know ;) |
| 15:44.35 | Talkradio | i actually called sagnoma for the a200 and they called me back the next day which was nice. it was a sunday i called them so i guess they don't do weekends for normal consumers |
| 15:45.03 | Katty | no one does weekends. |
| 15:45.11 | Katty | that's bad for business. |
| 15:45.35 | madgeek | ppl do weekends |
| 15:45.38 | madgeek | if you pay for it |
| 15:45.40 | Talkradio | does weekends to pay the bills plus no end users or traffic |
| 15:46.03 | FunkyGMT | If anyone can figure this out, I've a pretty weird problem with a phone. (SPA942). I've 2 asterisk servers, (1.2 & 1.4). The SPA942 is behind a NAT. It registers and working #1 when connecting on the 1.2, but doesn't receive any SIP response from the 1.4 on registering. It's veryyyy weird. |
| 15:46.36 | FunkyGMT | Both servers are public, not behind a Nat, and to make sure it was not related to it, I've closed firewalls. |
| 15:47.45 | Katty | Talkradio: it's usually just not productive for a big company to fully staff their building on a weekend. |
| 15:48.08 | Talkradio | oh yeah i totally agree |
| 15:50.38 | *** join/#asterisk eric256 (n=ejh@72.1.122.62) |
| 15:51.03 | eric256 | is there a good place with info about Asterisk Manager API? |
| 15:51.34 | eric256 | thanks for all the help over the last couple days btw ;) i now have a webpage that i can use to move any channel on the system and transfer it to a new extension ;) |
| 15:52.29 | Katty | jbot: api? |
| 15:52.29 | jbot | extra, extra, read all about it, api is (Application Programming Interface) APIs allow you to program to a pre-constructed interface (the API) instead of programming a device or piece of software directly. This allows for faster development, since programming to a device's API is designed to be easier than programming to a device directly. APIs allow you to program without having intimate knowledge of the device or software that you a |
| 15:52.42 | Katty | jbot: you're not very helpful sometimes. |
| 15:52.43 | jbot | no *you're* not very helpful sometimes. |
| 15:52.50 | eric256 | lmao |
| 15:52.55 | *** join/#asterisk LuisTorres (n=chatzill@81.84.224.208) |
| 15:53.09 | LuisTorres | Sorry just crashed |
| 15:53.19 | Katty | jbot: <3 |
| 15:53.20 | jbot | [<3] not >4, or not 3, or the emulation of a love symbol |
| 15:53.32 | [TK]D-Fender | lol |
| 15:53.33 | Katty | of course i'm not helpful. i'm a girl. |
| 15:53.35 | eric256 | been using http://www.voip-info.org/wiki/view/Asterisk+Manager+API+AgentLogoff but that specific command says its Invalid/Unknown...so i was hoping for another source |
| 15:53.57 | [TK]D-Fender | eric256: yup... |
| 15:53.59 | [TK]D-Fender | ~book |
| 15:54.00 | jbot | book is probably probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 15:54.01 | [TK]D-Fender | ~wikis |
| 15:54.02 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 15:54.26 | [TK]D-Fender | eric256: and in your source tarball. The WIKI is often outdated. |
| 15:54.31 | *** join/#asterisk ScarEye (n=scareye@12.27.87.66) |
| 15:55.00 | *** join/#asterisk mazpe (n=mazpe@c-76-109-233-122.hsd1.fl.comcast.net) |
| 15:55.17 | KavanS | Katty: fail |
| 15:55.27 | *** join/#asterisk Deeewayne (n=dwayne@nat/digium/x-90f9beab77754284) |
| 15:55.27 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:55.30 | Katty | you just wish you were a female. |
| 15:55.34 | Katty | hugs Deeewayne |
| 15:56.01 | KavanS | Katty: not really, just that excuse isn't valid...."I'm a girl"....for all we know you are a 45 year old WoW playing overweight man :) |
| 15:56.10 | Deeewayne | offers Katty a toasted and buttered bagel |
| 15:56.36 | Katty | KavanS: i can see you've never met me at a conference before... |
| 15:56.43 | Katty | Deeewayne: woohooo! |
| 15:56.48 | KavanS | lol most likely not.... |
| 15:56.48 | Katty | Deeewayne: <3 |
| 15:57.12 | Deeewayne | its bagel day @ digium! |
| 15:57.57 | Katty | i didn't know they had a bagel day |
| 15:58.05 | Katty | or is this a new cafe thing perk? |
| 15:58.30 | Katty | I can't believe they didn't pick my Little Star Coffee Bar name |
| 15:58.38 | Katty | oh well ;) |
| 15:59.28 | eric256 | [TK]D-Fender: thanks...turns out its now QueueRemove ;) |
| 15:59.41 | Deeewayne | Katty: no it is unrelated to Beans and Bytes |
| 15:59.43 | *** join/#asterisk wonderworld (n=ww@ip-62-143-20-187.unitymediagroup.de) |
| 16:00.50 | Katty | jaytee: do you get a bagel day? :< |
| 16:02.10 | *** join/#asterisk thansen (n=thansen@c-76-27-110-194.hsd1.ut.comcast.net) |
| 16:02.12 | Katty | jbot: isymphony? |
| 16:02.13 | jbot | i guess isymphony is http://www.i9technologies.com/ |
| 16:02.31 | Katty | jbot: does isymphony work with the new 1.6 api yet? |
| 16:02.49 | Katty | jbot: :< |
| 16:02.50 | jbot | < is, like, redirection of stdin to a program |
| 16:02.54 | Katty | jbot: look that up for me. thx. |
| 16:05.22 | *** join/#asterisk jks (n=jks@193.189.93.254) |
| 16:06.10 | Katty | someone in here works for i9 technologies |
| 16:06.14 | Katty | but i forget who |
| 16:09.26 | *** join/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/zeeek) |
| 16:09.32 | Katty | oh boy, it's Zeeek |
| 16:09.41 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 16:09.58 | Zeeek | Why Katty... how nice to see you again |
| 16:10.10 | Zeeek | oej neckties not allowed |
| 16:10.40 | [TK]D-Fender | Zeeek: Klaatu veratta.... necktie? |
| 16:10.43 | Zeeek | But you are all invited to join #voip-users-conference and talk to our guest from COunterpath live on the hour |
| 16:11.06 | Zeeek | Nicotine verrata framboise |
| 16:12.06 | Zeeek | after you ahve completed the IQ test by joining #voip-users-conference, you may want ot go look at http://tr.im/voip to see that various numbers and SIP URI available to you in wideband or howwlopoint |
| 16:12.15 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 16:12.35 | bmoraca | Katty: really? that's interesting... iSymphony's a really neat program |
| 16:13.05 | *** part/#asterisk KavanS (i=proxy@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 16:13.14 | Katty | bmoraca: baroo? what's interesting? |
| 16:13.25 | Katty | hugs zeek |
| 16:13.26 | Katty | oh |
| 16:13.27 | Zeeek | what? what? |
| 16:13.28 | Katty | hugs Zeeek |
| 16:13.31 | Katty | my alttab failed me. |
| 16:13.31 | bmoraca | that someone in here works for i9 |
| 16:13.32 | Katty | <PROTECTED> |
| 16:13.42 | Zeeek | snuggles up in his furry jacket |
| 16:14.55 | *** join/#asterisk verywiseman (n=verywise@unaffiliated/verywiseman) |
| 16:19.45 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 16:20.24 | Katty | bmoraca: you'd be surprised who shows up in here |
| 16:20.32 | Katty | bmoraca: riddlebox isn't too far away from me. i could probably meet him for lunch |
| 16:20.40 | Katty | bmoraca: and i've seen someone from my ISP randomly show up too |
| 16:21.01 | eric2 | I always am surprised too |
| 16:21.37 | Katty | if they'll let me in, they'll let anyone in ;) |
| 16:22.56 | Zeeek | in where Katty? |
| 16:23.05 | Katty | here |
| 16:23.17 | *** join/#asterisk fred-tmft (n=fred-tea@c-69-244-180-112.hsd1.mi.comcast.net) |
| 16:23.18 | Zeeek | the universe? |
| 16:23.25 | Katty | yes. |
| 16:23.28 | Katty | obviously. |
| 16:23.32 | *** part/#asterisk fred-tmft (n=fred-tea@c-69-244-180-112.hsd1.mi.comcast.net) |
| 16:23.37 | Zeeek | You are the universe I thought |
| 16:23.49 | Katty | only to me!!! *hee* |
| 16:23.51 | *** join/#asterisk mechbangirc (n=mech@119.153.31.231) |
| 16:24.07 | Katty | does this universe make my thighs look big? |
| 16:24.24 | Zeeek | only when you turn a certain way |
| 16:25.27 | Zeeek | the people who create some of these spam subjects could be in legit advertising |
| 16:25.31 | *** join/#asterisk deadpigeon (n=deadpige@office.xpressamerica.net) |
| 16:26.52 | riddlebox | Katty lets meet and plot our takeover of this area!! |
| 16:26.56 | Katty | i got some fun spam this morning |
| 16:26.58 | Zeeek | Join the VUC on #voip-users-conference - if you have wideband SIP client you can call the wideband conference bridge - see you over there |
| 16:27.10 | Katty | riddlebox: you can just have it lol |
| 16:27.15 | Katty | riddlebox: i wanna go work at a zoo |
| 16:27.32 | riddlebox | the hairless chimp died here |
| 16:28.02 | Katty | brb |
| 16:28.03 | mechbangirc | hi i setup ooh323.conf how do I match remote party id. do I need their H323ID, cause it is not matching on ip |
| 16:28.04 | Katty | afks |
| 16:29.07 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 16:29.22 | Kobaz | mechbangirc: if you get h323 going, i would like to know |
| 16:29.54 | mechbangirc | call gets connected but no voice either way and then dc |
| 16:30.02 | Kobaz | yeap |
| 16:30.04 | Kobaz | that's what i get |
| 16:31.10 | jmkgreen | Anyone have any idea why Asterisk thinks three out of four PSTN handsets have hung up dispite that not being the case? |
| 16:31.16 | mechbangirc | i dont understand H323 technology. SIP and IAX work great |
| 16:31.51 | jmkgreen | this is now happening repeatedly |
| 16:31.57 | *** part/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/zeeek) |
| 16:32.06 | Qwell | jmkgreen: any more details? |
| 16:32.12 | jmkgreen | small number of outbound test calls and we're seeing user hangups on some of them |
| 16:32.34 | jmkgreen | they are all in the office and we know asterisk is hanging up itself |
| 16:32.41 | jmkgreen | the logs show a user hangup however |
| 16:33.08 | jmkgreen | I'm not sure how many other details I can give !:) |
| 16:33.51 | jmkgreen | we're seemingly up against a brick wall - we are sending the calls out via two different voip providers so it's not their fault either |
| 16:33.54 | Qwell | show us the logs |
| 16:33.58 | Qwell | ~pastebin |
| 16:33.59 | jbot | [~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 16:34.01 | jmkgreen | there's a lot of logs |
| 16:34.27 | mechbangirc | hereis my ooh323.conf http://pastebin.ca/1342965 |
| 16:34.37 | mechbangirc | brb in 10 min |
| 16:34.56 | jmkgreen | Qwell: I'm told the logs have no errors at all, just a user hangup |
| 16:35.02 | Qwell | show us the logs |
| 16:35.04 | jmkgreen | we're at verbosity 15 |
| 16:35.19 | jmkgreen | engineer's checking the iax2 logs now |
| 16:35.23 | SuPrSluG | hello all |
| 16:37.02 | SuPrSluG | voicemails stop recording after a minute on inbound calls only. what causes this? |
| 16:38.17 | jmkgreen | Qwell: About to have the pastbin link |
| 16:39.00 | jmkgreen | Qwell: http://papernapkin.org/pastebin/view/4532/ |
| 16:39.29 | jmkgreen | That's one call which came to an office telephone and hung up once we'd answered with no audio played to us |
| 16:39.35 | Qwell | vxml? |
| 16:39.39 | *** join/#asterisk brunner (n=chris@68-119-87-106.dhcp.mtgm.al.charter.com) |
| 16:39.56 | jmkgreen | Yes we're using voicexml to prerecord messages to be played |
| 16:39.59 | *** join/#asterisk thehar (i=thehar@thehar.xmission.com) |
| 16:40.03 | brunner | can agi-bin be a symbolic link? |
| 16:40.45 | Qwell | jmkgreen: your problem is in there |
| 16:40.58 | jmkgreen | In where? |
| 16:42.16 | jmkgreen | the vxml? |
| 16:42.33 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com) |
| 16:42.36 | jmkgreen | it is catching a user hangup, logging it to a remote http server, then exiting |
| 16:43.02 | jmkgreen | but this system has worked fine in the past, indeed some calls are getting through and playing their audio fine |
| 16:43.23 | jmkgreen | we've had eight simultaneous calls played without problem |
| 16:43.29 | Qwell | The problem is in there.. Asterisk isn't seeing a hangup |
| 16:43.42 | jmkgreen | Asterisk isn't seeing a hangup? |
| 16:44.00 | Qwell | Do you see Asterisk saying anything about a hangup before it exits vxml? |
| 16:45.17 | rene- | vxml exits before |
| 16:45.28 | rene- | vxml is making asterisk hangup |
| 16:45.56 | rene- | is the same with other similar systems such as agi |
| 16:46.07 | Qwell | and it's highly unlikely that anybody in here will be able to help you debug that, since it's a third party app |
| 16:46.11 | rene- | a logical syntax error or a condition in your external script |
| 16:46.27 | rene- | is being met and cause to abruptly end the call |
| 16:49.12 | *** join/#asterisk jcoffi (n=jcoffi@190.213.22.5) |
| 16:49.18 | jmkgreen | well you've got the engineer scratching his head now. No changes to the vxml since it was working yesterday so he assumed asterisk was receiving the hangup from the voip provider I think |
| 16:49.26 | *** join/#asterisk wacky_ (n=abourget@IP-208-88-110-46.mtl.fibrenoire.ca) |
| 16:49.34 | wacky_ | is there a way to lower the volume of the music on hold ?? |
| 16:49.41 | Qwell | jmkgreen: look at the timestamps on the log |
| 16:49.42 | jmkgreen | he asks what asterisk should report in it's logs when the user hangs up? |
| 16:49.50 | Qwell | 1 second for all of that |
| 16:51.01 | Qwell | ~book |
| 16:51.01 | jbot | book is, like, probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 16:51.15 | Qwell | jmkgreen: Tell your engineer to get reading. |
| 16:51.17 | *** join/#asterisk telnettech (i=telnette@gw.percipia.com) |
| 16:51.57 | mechbangirc | i have problem with ooh323.conf |
| 16:52.03 | jmkgreen | apparently he's changed the polarity setting? because we were not detecting hangups properly |
| 16:52.21 | Qwell | so, when he says nothing, he means a critical setting? |
| 16:52.24 | Qwell | got it. |
| 16:52.32 | Qwell | never getting those 10 minutes of my life back. |
| 16:53.21 | Qwell | gee...a problem starts when you change a setting. I WONDER WHAT CAUSED THE PROBLEM |
| 16:53.35 | *** join/#asterisk jcoffi1 (n=jcoffi@75.147.157.49) |
| 16:53.36 | madgeek | erich is that you? |
| 16:53.47 | madgeek | sorry Qwell, reminded me of a guy i work with |
| 16:54.01 | jmkgreen | apparently the setting was changed on the 10th of the month and there's not been such problems until today |
| 16:54.17 | Qwell | jmkgreen: make him read the book |
| 16:54.32 | mechbangirc | http://pastebin.ca/1342977 calls connect and then disconnect without any voice from either side |
| 16:55.01 | *** join/#asterisk jad_jay (n=chatzill@public.axolys.fr) |
| 16:55.18 | mechbangirc | i am getting a warning "Don't know how to indicate condition 20 on ooh323c_34" |
| 16:55.23 | *** join/#asterisk freakazoid0223 (n=matt@pool-71-246-17-74.phlapa.fios.verizon.net) |
| 16:56.06 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) |
| 16:57.21 | *** part/#asterisk steerpike (n=Unknown@unaffiliated/steerpike) |
| 16:58.35 | mechbangirc | followed by an error saying "exited non-zero on 'OOH323/(null)-8e58'" |
| 17:02.16 | *** join/#asterisk stuffcorpse (n=rick@121.98.136.241) |
| 17:05.48 | *** join/#asterisk erth64net (n=erth64ne@69-30-67-191.dq1sn.easystreet.com) |
| 17:13.05 | *** join/#asterisk hatoon (n=ujzfwwop@pontanegra.act.psi.br) |
| 17:13.14 | *** part/#asterisk hatoon (n=ujzfwwop@pontanegra.act.psi.br) |
| 17:14.16 | madgeek | http://news.bbc.co.uk/2/hi/europe/7900154.stm |
| 17:14.19 | madgeek | ooops |
| 17:14.24 | madgeek | wrong chan |
| 17:18.19 | *** join/#asterisk freckle_home (n=chatzill@195.74.96.119) |
| 17:19.32 | *** part/#asterisk rpm (n=rpm@S010600055d2cf2e2.cg.shawcable.net) |
| 17:22.37 | *** join/#asterisk tacubo (n=tacvbo@189.146.190.196) |
| 17:23.42 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 17:23.56 | Whitor | <PROTECTED> |
| 17:24.07 | Whitor | <PROTECTED> |
| 17:24.33 | Whitor | 4 is the start of an extension at the remote asterisk box |
| 17:24.57 | thansen | any celliax experts around? |
| 17:25.12 | Whitor | <PROTECTED> |
| 17:26.13 | *** join/#asterisk lanning (n=lanning@173.8.187.197) |
| 17:26.32 | Corydon76-dig | Whitor: what exactly were you expecting? |
| 17:27.19 | SuPrSluG | adding an option to asterisk.conf . does that require a restart or will a reload do? |
| 17:27.49 | Whitor | Corydon76-dig, I was expecting that it would allow me to enter three digits (for an extension) then perform a lookup... see that they weren't local, perform a dundi lookup, find them on the remote system, and trunk the call |
| 17:28.15 | Corydon76-dig | Whitor: do you have a switch statement in that context referencing dundi? |
| 17:29.14 | *** join/#asterisk manxpower (n=Administ@router.asteriasgi.com) |
| 17:29.26 | Whitor | Corydon76-dig, I'm following a guide I found online ... So, honestly, I'm still learning all the lingo |
| 17:29.54 | brunner | in sip.conf, how can I make two different sip accounts go to two different contexts? according to voip-info.org, "context" is not a valid variable for peer definitions. |
| 17:30.15 | manxpower | brunner: voip-info is wrong (it usually is) |
| 17:31.01 | Whitor | Corydon76-dig, How do I check if I have a switch statement referencing that context? |
| 17:31.26 | manxpower | Technically, in 1.2 and earlier context was not valid in a peer statement (peers accept calls from asterisk so why would you be able to specify a context). |
| 17:31.33 | Whitor | err... How do I check if I have a switch statement in that context referencing dundi ... |
| 17:31.43 | manxpower | In 1.4 and later peers are really treated as friends, so you could put a context= in there. |
| 17:32.01 | manxpower | Whitor: um, look at extensions.conf? |
| 17:33.08 | *** join/#asterisk imchandave-BB (n=imchanda@fw-e.isp.sunday.com) |
| 17:33.32 | brunner | actually, I can't even get asterisk to register with my sip proxy this way |
| 17:33.42 | brunner | I'm sure I'm doing something really stupid. would someone take a glance? http://pastebin.com/m1e2102f7 |
| 17:34.21 | Whitor | manxpower, I have that context in extension_custom.conf |
| 17:34.39 | Whitor | *extensions_custom.conf |
| 17:35.56 | Qwell | drmessano: well? |
| 17:36.21 | brunner | I guess everything in [general] doesn't automatically carry and must be redefined for each peer definition, huh? |
| 17:36.51 | Whitor | This is my extensions_custom.conf http://pastebin.ca/1343023 |
| 17:36.53 | manxpower | brunner: no, not everything, there are a few things that don't carry over |
| 17:37.05 | manxpower | Whitor: You asked a question, I answered. |
| 17:37.06 | Corydon76-dig | brunner: not everything, but you can define templates |
| 17:37.31 | manxpower | Whitor: sounds to me like you are using a GUI |
| 17:37.51 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 17:37.51 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 17:37.51 | *** mode/#asterisk [+o russellb] by ChanServ |
| 17:38.00 | brunner | manxpower, Corydon76-dig: well would you take a look at my sip.conf and tell me what I should be defining in my peer definition? because it all works when everything is in general: http://pastebin.com/m1e2102f7 |
| 17:38.02 | Whitor | manxpower, at this point, I am doing everything cli |
| 17:38.22 | Whitor | manxpower, I appreciate your answer |
| 17:38.41 | manxpower | brunner: register => is not valid anywhere except for [general] |
| 17:38.48 | manxpower | What piece of crap documentation made you think it was? |
| 17:39.14 | telnettech | i have a question about Polycom 650 with a sidecar......I am trying to get the contact directory to be speed dials on the sidecar. I see that the sidecar has power and appears to be connected correctly.....there is entries in the contact directory of the phone that I am able to use the menu to get to.....any idea what I may be missing configuration wise |
| 17:39.21 | Whitor | How do I get one system to allow entering of a remote system's extensions during the IVR? |
| 17:39.23 | brunner | manxpower: damn, well then how do I register different sip accounts under different contexts? |
| 17:39.51 | Corydon76-dig | brunner: If it works, what are you complaining about? |
| 17:39.58 | manxpower | brunner: Registration notifies the remote system what the ip address is of a user/secret. It does nothing else. Let me repeat, it does NOTHING ELSE> |
| 17:40.09 | brunner | Corydon76-dig: it doesn't work |
| 17:40.22 | Corydon76-dig | brunner: see what manxpower just said |
| 17:40.22 | brunner | asterisk won't register with the sip proxy when it's done likethis |
| 17:40.23 | manxpower | once the remote system knows your IP then calls come in to the user/friend/peer entry that matches the incoming call. |
| 17:40.49 | manxpower | brunner: doing it the way you are doing it right now is guaranteed to fail. |
| 17:40.50 | Corydon76-dig | brunner: the secret MUST be defined separately |
| 17:41.07 | Corydon76-dig | brunner: as should the username |
| 17:41.46 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
| 17:42.00 | brunner | okay, got it. thanks. |
| 17:42.25 | *** join/#asterisk Khratos (n=khratos@190.166.103.227) |
| 17:42.26 | SuPrSluG | telnettech:you configured <mac>-directory.xml ? |
| 17:43.42 | telnettech | SuPr: I have a .xml file and it displays on the phone itself thru the contact directory....the porblem is that they dont show on the sidecar.....there are speed dial indexes showing in the .xml file |
| 17:44.12 | manxpower | telnettech: I have never heard of anyone with that problem. |
| 17:44.39 | telnettech | manx: I have resetted the phone and verified that the sidecar rboots with the phone |
| 17:44.41 | manxpower | the speed dials start at the first unconfigured line appearance or side card key |
| 17:44.49 | Khratos | good `date +%r` |
| 17:45.18 | telnettech | and there is nothing on the config of the phone to make it know that there is a sidecar right? |
| 17:45.34 | manxpower | I have a 650 w/side car on my desk. |
| 17:45.36 | manxpower | no problems. |
| 17:45.45 | SuPrSluG | no it just recognizes the sidecar |
| 17:45.57 | telnettech | i have 1 too with no problems also but the customer has 1 and it is not working |
| 17:46.21 | telnettech | like i said the contact directory shows on the phone but the siecar is blank |
| 17:46.26 | manxpower | telnettech: what version of the sip firmware do you have on that phone? |
| 17:47.20 | telnettech | let me check |
| 17:48.02 | hardwire | hmm.. any way to round robin calls through two zap groups? |
| 17:48.22 | manxpower | hardwire: assign the channels to two groups |
| 17:48.24 | hardwire | just set an extension variable.. check for odd via division.. then use a goto? |
| 17:49.10 | hardwire | manxpower: then using round robin it goes through all 24 channels of span1 before going to span2 |
| 17:49.29 | manxpower | hardwire: whatever you want. |
| 17:49.37 | hardwire | I even set: |
| 17:49.46 | hardwire | group=3 |
| 17:49.47 | hardwire | channel => 25 |
| 17:49.47 | hardwire | channel => 73 |
| 17:49.47 | hardwire | channel => 26 |
| 17:49.47 | hardwire | channel => 74 |
| 17:49.47 | hardwire | ... |
| 17:49.53 | hardwire | but it sorted them :) |
| 17:50.15 | manxpower | hardwire: A channel can be a member of more than one group. Think about that for a moment. |
| 17:50.35 | hardwire | using group=2,3 ? |
| 17:50.41 | manxpower | hardwire: correct. |
| 17:50.53 | hardwire | ah |
| 17:50.59 | hardwire | I was overriding it. |
| 17:51.25 | hardwire | manxpower: either way.. it still sorts the channels in each group |
| 17:51.42 | manxpower | hardwire: :sort" |
| 17:51.46 | manxpower | it does nothing of the sort. |
| 17:51.50 | hardwire | hah |
| 17:51.54 | hardwire | I could have sworn I just tested it |
| 17:51.57 | hardwire | lemme try gain yo. |
| 17:52.03 | manxpower | groups can be referenced as G, g, R, or R |
| 17:52.07 | manxpower | pick which one you want. |
| 17:52.41 | hardwire | I'm using r |
| 17:53.43 | manxpower | g1 = use lowest numbered avail channel. G1 = use highest numbered channel r1 = use the next available channel with a number higher than the last used channel |
| 17:53.46 | hardwire | when asterisk loads zapata.conf it appears to ignore ordering in the conf file and simply relates channels to groups. |
| 17:53.48 | manxpower | R1 would be the reverse |
| 17:53.50 | hardwire | manxpower: ... |
| 17:54.02 | hardwire | yessir.. I'm aware of that, even though it doesn't hurt to point it out. |
| 17:54.12 | manxpower | r/R has a MEMORY. g/G does NOT have a memory. |
| 17:54.35 | eric256 | wouldn't it follow then that if you are using r it would go in order? lowest to highest? |
| 17:54.50 | manxpower | eric256: sort of. |
| 17:55.16 | eric256 | well yea sort of ;) |
| 17:55.35 | manxpower | Lets say channels 1, 4, 8 are in use. the last channel that was active is channel 20. r1 would use channel 21 even thought there are lots of lower numbered channels available |
| 17:55.58 | manxpower | If you want it to always use the lowest numbered avail channel then use r1 |
| 17:56.08 | hardwire | manxpower: right now I have channels 25-48 and 73-96 in group 2 |
| 17:56.16 | hardwire | and I have interleaved them into group 3 as well |
| 17:56.27 | hardwire | 25,73,26,74,... |
| 17:56.31 | manxpower | hardwire: interleaved |
| 17:56.32 | telnettech | manx: the sip firmware is 2.1.0.2708 |
| 17:56.48 | hardwire | when I use r3 as the group name it goes from 25.. to 26 |
| 17:57.06 | hardwire | when I go all the way to 48 it goes to 73 |
| 17:57.41 | Qwell | hardwire: I don't think you can define groups like that |
| 17:57.48 | Qwell | like, a channel in multiple groups |
| 17:57.50 | hardwire | Qwell: that's what I'm trying to establish. |
| 17:57.52 | Qwell | maybe, but... |
| 17:58.06 | Qwell | I suspect if you have group= twice for one channel, it'll only use one |
| 17:58.14 | manxpower | telnettech: sounds like you are using a 2.0.x sip.cfg/phone1.cfg with 2.1 SIP firmware with the 650. That won't work |
| 17:58.23 | telnettech | ok |
| 17:58.24 | manxpower | Qwell: that is my experience. |
| 17:58.34 | hardwire | Qwell: commas ;) |
| 17:58.44 | manxpower | hardwire: put your zapata.conf on pastebin.ca |
| 17:58.50 | Qwell | I'm pretty sure it uses an sscanf :p |
| 17:58.59 | Qwell | which means it'll just take the first number in the statement |
| 17:59.20 | manxpower | Qwell: group=1,3 channel => 1-24 is valid and will work and the channels are in both groups. |
| 17:59.30 | Qwell | oh? |
| 17:59.57 | manxpower | Qwell: Yup, has worked since at least 1.2 |
| 18:00.06 | hardwire | manxpower: http://pastebin.ca/1343044 |
| 18:00.30 | Qwell | ah, see, you're redefining channels there |
| 18:00.46 | hardwire | Qwell: it's just a test, but yes |
| 18:00.49 | *** join/#asterisk propellerhead (n=yogurt2u@host43.190-137-13.telecom.net.ar) |
| 18:00.51 | manxpower | hardwire: stop having more than one group line for the same channel! |
| 18:01.07 | Qwell | try more like... |
| 18:01.19 | hardwire | manxpower: ok.. and? if I comment them out it will do nothing to how asterisk uses it. |
| 18:01.30 | Qwell | group=1,2 \n channel => 25 \n group=1 \n channel => 26-48 |
| 18:01.31 | hardwire | the last bit IS an override. |
| 18:01.40 | manxpower | I'll rewrite your file for you |
| 18:02.30 | manxpower | hardwire: take my advice or don't take it. I have given my advice. |
| 18:02.41 | hardwire | I'm not refusing it. |
| 18:03.09 | manxpower | then redo your file so you do not have all those extra group= and channel= lines in it. |
| 18:03.11 | hardwire | I'm interested to see how you would write this, keep in mind I assumed I was overriding groups for channels |
| 18:03.31 | manxpower | hardwire: exactly how Qwell just said |
| 18:04.13 | Qwell | though if all of those channels are in groups 2 and 3, why not just set group=2,3 instead of group=2? |
| 18:04.14 | hardwire | I still haven't established one thing. |
| 18:04.26 | hardwire | will rr call each channel in the order it is described in zapata.conf ? |
| 18:04.46 | hardwire | I don't know if there is an answer to that in everything everybody has been telling me. |
| 18:05.03 | manxpower | lets say I have channels 1-23, I want user dialed calls to start at the highest numbered channel working to lowest and I want the voicemail notify script to only use channels 1 - 16. group=1,2 \n channel => 1-16 \n group=1 \n channel 17-23 |
| 18:05.21 | manxpower | user calls would be Dial(Zap/G1/whatever) |
| 18:05.33 | manxpower | the voicemail notifiy would be Dial(Zap/G2/whatever) |
| 18:05.57 | hardwire | right because you are overriding the group to simply be 1 for 17-23 |
| 18:06.01 | hardwire | similar to what I'm doing. |
| 18:06.04 | eric256 | hardwire: manxpower said..."r1 = use the next available channel with a number higher than the last used channel"... i'm pretty sure that answers your question doesn't it? |
| 18:06.23 | hardwire | eric256: thanks.. so it's sorted |
| 18:06.26 | hardwire | much appreciated |
| 18:06.32 | hardwire | finds another way to do what he wants. |
| 18:06.37 | manxpower | hardwire: they are not sorted at all! |
| 18:06.41 | manxpower | they are rmembered |
| 18:06.43 | hardwire | +1 = sort |
| 18:06.46 | manxpower | no. |
| 18:06.53 | manxpower | sort rearranges the order of things. |
| 18:06.56 | hardwire | 25+1=26 |
| 18:06.58 | manxpower | R/r does nothing of the sort. |
| 18:07.11 | manxpower | r1 means use one less than the last used channel |
| 18:07.21 | manxpower | it does not mean "reorder the channels" |
| 18:07.29 | hardwire | manxpower: right.. if the last channel was zap25 then the next will be zap26 no matter how I defined them in zapata.conf |
| 18:07.44 | hardwire | it +1's until it finds an available channel in that group |
| 18:07.49 | hardwire | that is inherently, a sort. |
| 18:08.03 | manxpower | hardwire: um, order of definition does not matter no matter what |
| 18:08.26 | hardwire | heh |
| 18:08.38 | hardwire | anyways.. much appreciated |
| 18:08.49 | hardwire | thanks eric256. |
| 18:09.55 | *** join/#asterisk RobertLaptop (n=rmiddle@m9a0736d0.tmodns.net) |
| 18:10.42 | watchy2 | anyone know the maximum distance a phone could be from a Mediatrix? |
| 18:16.17 | *** join/#asterisk mapmagic (n=piyush@chan.cs.fsu.edu) |
| 18:16.18 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 18:16.33 | hardwire | that makes me curious as to what your current length is |
| 18:16.39 | mapmagic | if i hook up a phone on my internet using voip, is there a way to dial 911? |
| 18:16.58 | hardwire | mapmagic: if your VoIP provider supports it, yes. |
| 18:17.08 | hardwire | it's usually a costs a bit more feature. |
| 18:17.24 | mapmagic | any recommendations for a voip provider that supports decent rates both inside the US and outside + 911 support? |
| 18:17.48 | hardwire | afaik broadvoice groks e911.. teliax as well. |
| 18:18.23 | manxpower | mapmagic: 911 is easy. e911 is not supported by everyone. |
| 18:18.38 | manxpower | "e" = provides your location to 911 service. |
| 18:21.10 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 18:21.53 | *** join/#asterisk TheIceMAn (n=theicema@86.122.46.5) |
| 18:22.41 | TheIceMAn | i am having a problem with asterisk 1.5 beta, asterisk is not always closing the outside line, can that be from some parameters ? |
| 18:23.38 | manxpower | TheIceMAn: Odd since there has never been an Asterisk 1.5 beta. |
| 18:24.25 | TheIceMAn | asterisk now |
| 18:24.26 | TheIceMAn | :D |
| 18:25.16 | *** part/#asterisk jcoffi (n=jcoffi@75.147.157.49) |
| 18:25.25 | manxpower | TheIceMAn: Then call it Asterisk NOW |
| 18:25.33 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 18:25.34 | *** join/#asterisk Juggie (n=Juggie@CPE001601df17fb-CM001ceac25ada.cpe.net.cable.rogers.com) |
| 18:25.36 | TheIceMAn | sorry |
| 18:25.48 | TheIceMAn | Dial("SIP/326-09a5d720", "DAHDI/g1||tr") |
| 18:25.51 | TheIceMAn | is this okey ? |
| 18:25.57 | manxpower | Why is this orange red? Because it's an apple. |
| 18:26.01 | TheIceMAn | or should i remove tr ? |
| 18:26.34 | manxpower | I can't comment on AsteriskNOW |
| 18:27.01 | manxpower | Last time I came across an Asterisk now machine I drew a pentagram around it, called for the clergy and burned it. |
| 18:27.03 | TheIceMAn | its has asterisk 1.4.22 on it |
| 18:28.14 | *** join/#asterisk n3hxs (n=HAMming@63.68.135.4) |
| 18:28.39 | mapmagic | I need some help. An organization that I work for needs voip to be setup for roughly 50 people. We want it to be run using asterisk and our internet (which is 2Gb/sec line). Can anyone here do this for us and charge us a one time cost? |
| 18:29.17 | manxpower | mapmagic: contact me via /msg. My employer does some of that sort of thing. |
| 18:29.40 | *** join/#asterisk AndyML (n=quassel@pool-98-111-140-229.phlapa.fios.verizon.net) |
| 18:29.52 | Kobaz | mapmagic: what's your budget? |
| 18:31.51 | AndyML | I have a "Flex" PRI through a tier1 provider that is giving me weird behavior through asterisk. When I call out to my cellphone from Asterisk through the Flex PRI, I get one-way audio (VoIP phone plays audio from cellphone callie) and no audio on the cellphone from the VoIP phone. Then, when everyone hangs up, the cellphone has a voicemail and it contains the VoIP/PRI side of the conversation. |
| 18:35.07 | *** join/#asterisk IPPBX-ARG (n=pirruar@190.176.157.156) |
| 18:35.26 | rene- | what is a flex pri anyway? |
| 18:35.38 | rene- | is it a voice/data PRI ? |
| 18:35.57 | [TK]D-Fender | rene-: Basically VoiP to an IAD that spits out PRI |
| 18:36.20 | [TK]D-Fender | rene-: Single data link and bandwidth it controlled to your provider |
| 18:36.20 | rene- | huh, i have one and i didnt even know what is called |
| 18:37.17 | eppigy | hello |
| 18:37.23 | rene- | hey |
| 18:38.28 | Katty | allo |
| 18:40.04 | eppigy | OI |
| 18:40.54 | *** join/#asterisk jcoffi1 (n=jcoffi@208.87.0.146) |
| 18:42.30 | Kobaz | i wonder if there's such a think as a usb fxo/fxs dongle... or even a usb t1 module |
| 18:42.59 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 18:44.12 | *** part/#asterisk c0rnoTa (n=c0rnoTa@80.251.113.56) |
| 18:44.37 | AndyML | [TK]D-Fender: ever seen such nonsense as split channels from a PRI? half the call (and the audio) goes to voicemail, and the other half goes to the user. |
| 18:45.17 | [TK]D-Fender | AndyML: No, you're on crack.... |
| 18:45.34 | [TK]D-Fender | AndyML: pastebi something worthwhile to look at. |
| 18:45.36 | mapmagic | manxpower: Thanks a lot. Dave was very helpful. |
| 18:45.49 | mapmagic | Any recommendations for wireless handset that works with asterisk and wifi? |
| 18:45.49 | AndyML | give me your cellphone number, i'll call you and talk to you AND leave you a voicemail at the same time. |
| 18:45.56 | mapmagic | cost is not a concern. |
| 18:46.39 | Kobaz | AndyML: sounds like you're dialing out two channels at the same time... ie Dial(zap/g0&zap/g1) |
| 18:47.08 | AndyML | it does doesn't it... I think the telco has their flex router configured wrong but they're blaming asterisk |
| 18:47.23 | *** join/#asterisk drudge` (i=anonymou@unaffiliated/drudge/x-837452) |
| 18:47.31 | *** join/#asterisk tobias (n=tobias@cpe-069-134-127-101.nc.res.rr.com) |
| 18:48.24 | drudge` | anyone use a voip phone that flashes a light or something to indicate a call is recording? we use asterisk |
| 18:48.33 | rene- | Kobaz: usb t1 does exist |
| 18:48.34 | eppigy | AndyML: what are you currently paying for an ounce of hard? Maybe we can work something out |
| 18:48.37 | rene- | google for xorcom |
| 18:48.44 | Kobaz | interesting |
| 18:49.10 | AndyML | eppigy: an ounce of hard? they're paying me, why? |
| 18:49.29 | rene- | hehe |
| 18:50.03 | rene- | i didnt realize that asterisk is all about connecting |
| 18:51.59 | *** part/#asterisk drudge` (i=anonymou@unaffiliated/drudge/x-837452) |
| 18:54.54 | [TK]D-Fender | rene-: USB isn't a T1 carrier medium. They can try to port the functional protocol to something that carries over it though. |
| 18:55.03 | [TK]D-Fender | rene-: its a question of encapsulation. |
| 18:55.56 | hardwire | anybody set up a simple global flipflop using astdb or global vars? |
| 18:55.56 | rene- | D-Fender: got it, so they never could actually run T1 frames over USB right |
| 18:56.25 | Kobaz | well it has to be encapsulated since usb has it's own protocol |
| 18:56.32 | tzafrir_laptop | [TK]D-Fender, if USB isn't, then ethernet cetainly isn't |
| 18:57.13 | rene- | iam no EE i just thought he wanted to terminate to a voice t1 over a usb port |
| 18:57.23 | *** join/#asterisk jeffgus (n=jeffgus@green.zimage.com) |
| 18:57.32 | Kobaz | yeah, i was just interested to see if they existed and if it was worth it price wise |
| 18:57.37 | tzafrir_laptop | USB is much nicer than ethernet |
| 18:57.55 | rene- | i would like to build POE switches |
| 18:58.04 | rene- | <PROTECTED> |
| 18:58.09 | Kobaz | ~itsp |
| 18:58.10 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 18:58.11 | hardwire | :) |
| 18:58.24 | Kobaz | ~itsplist-us |
| 18:58.24 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 18:58.52 | rene- | hey thats cool |
| 18:59.11 | tzafrir_laptop | [TK]D-Fender, in what way USB isn't functional that PCI is? |
| 19:00.01 | Kobaz | usb would be plenty fast enough as well, usb2 is more then plenty |
| 19:00.28 | AndyML | http://pastebin.com/m29e92a48 |
| 19:01.28 | AndyML | on that call, as soon as I dialed, I got my cellphone voicemail greeting, simultaneously I got the audio from the call ringing. When my cellphone rang, I hit "ignore" and the the IP phone started playing my voicemail greeting a second time - still on top of the first one. |
| 19:01.36 | AndyML | that is a pri intense debug |
| 19:02.28 | Kobaz | AndyML: your provider has something seriously screwed up |
| 19:03.12 | AndyML | yeah, that becomes more and more evident the more i debug this thing. Frankly I was hoping someone had see this before, whether or not the cause is within asterisk - these are common services now... |
| 19:05.15 | mazpe | how can i add a 1 onfront of a call dialed with 10 digits? |
| 19:07.00 | riddlebox | Dial(1{EXTEN}) |
| 19:07.07 | riddlebox | pretty much |
| 19:07.10 | jjshoe | god I <3 sangoma support |
| 19:07.22 | mazpe | simple |
| 19:07.26 | mazpe | thanks riddlebox |
| 19:09.38 | eppigy | hi |
| 19:09.41 | eppigy | i am dave |
| 19:12.42 | *** join/#asterisk exothermc (n=miles@74.85.89.146) |
| 19:13.03 | exothermc | is there anyway to get asterisk to proxy g.729 without paying for licenses? |
| 19:13.34 | exothermc | without reinviting the audio |
| 19:13.49 | *** join/#asterisk BBMitch (n=Miranda@66.199.170.249) |
| 19:15.46 | *** join/#asterisk fprior78 (n=chatzill@host105.200-43-137.telecom.net.ar) |
| 19:15.51 | BBMitch | Good morning from Vancouver Canada (GMT-8) - does anyone know where the difference between s, _., and _X, are documented? I am trying to debug the change in call handling on unrecognized between two versions - one works as desired, the new one catches all calls - even ones not desired - we are having trouble correcting the behaviour with the custom scripts generated by freePBX and thought a better |
| 19:15.51 | BBMitch | <PROTECTED> |
| 19:16.20 | BBMitch | We have some places in dialplan where noop works and some where noop does not seem to output anything... Thanks for your help! |
| 19:16.26 | Carlos_PHX | _. should never be used. |
| 19:16.35 | Carlos_PHX | _X. should match all not otherwise matched. |
| 19:17.15 | Carlos_PHX | s is used where you don't need any pattern matching, such as a macro. |
| 19:17.59 | fprior78 | Hi all from Argentina. I'm using asterisk with LinkSys SPA400. When I generate a outbound call asterisk execute Answer() method immediately, during SPA400 dialing number. Can anyone help me ? |
| 19:18.05 | [TK]D-Fender | BBMitch: Documented in the source tarbal (you never downloaded), in the BOOK, on th4e WIKI, etc. |
| 19:18.07 | [TK]D-Fender | ~book |
| 19:18.08 | jbot | i guess book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 19:18.09 | [TK]D-Fender | ~wikis |
| 19:18.10 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 19:18.51 | exothermc | might want to update that wiki entry. |
| 19:19.25 | *** join/#asterisk JerJer (n=PhatJ@24-236-207-64.dhcp.aldl.mi.charter.com) |
| 19:19.29 | [TK]D-Fender | exothermc: Works fine for me.. |
| 19:20.05 | exothermc | There is SER and openSIPs and then that other one kam... that now looks like it is going to be going back to SER since all the devs moved to opensips. |
| 19:21.02 | EmleyMoor | Is there a way to send a Jabber message if a voicemail was left, or a voicemail count when a user connects to Jabber? |
| 19:21.48 | [TK]D-Fender | EmleyMoor: Yes, three are clear scripting hooks in your VM config. The latter is entirely up to you |
| 19:22.07 | *** join/#asterisk jasonwoot (n=some@bookit-dev.com) |
| 19:23.57 | jasonwoot | do any iSymphony users know how to kick of a rebuild of extensions.xml? |
| 19:28.57 | exothermc | [TK]D-Fender: Can Asterisk proxy g.729 audio without needing a license? |
| 19:29.08 | [TK]D-Fender | exothermc: * is NOT A PROXY |
| 19:29.23 | [TK]D-Fender | exothermc: * can pass on G.729 anywhere it doesn't need to trancode |
| 19:29.42 | exothermc | [TK]D-Fender: without a license? |
| 19:30.06 | [TK]D-Fender | exothermc: Thats what I just said |
| 19:33.19 | BBMitch | Wow - thanks Carlos - so here's by confusion then... I see _. used in the [ext-did-catchall] section as written by FreePBX - there is a set of exten lines from s1... sn followed by lines that are _.,1.... (I don't want to paste here) - so is that FreePBX generating lines they should not? And is that my problem do you think? |
| 19:40.26 | hardwire | lul.. I am the master of awesome! |
| 19:40.41 | hardwire | I should add my recording scenario to the wiki at some point |
| 19:40.45 | hardwire | but it just keeps getting cooler. |
| 19:46.37 | *** join/#asterisk voipforces (n=courchea@67.55.25.221) |
| 19:47.45 | justdave | anyone know if it's possible to build the dahdi echo cancelation drivers on RHEL5? |
| 19:47.59 | justdave | the instructions say to grab source from an upstream kernel, but it's a way newer kernel than the one RHEL uses |
| 19:49.16 | [TK]D-Fender | justdave: jsut install like normal |
| 19:49.26 | *** part/#asterisk exothermc (n=miles@74.85.89.146) |
| 19:49.29 | [TK]D-Fender | justdave: you need the sources and headers for YOUR kernel |
| 19:49.45 | justdave | my kernel doesn't have the echo cancelation driver though |
| 19:50.00 | justdave | the readme says they removed it from dahdi because it's included in upstream kernels now |
| 19:51.20 | voipforces | Hi all, I have 2 * server linked via IAX. anyway for server A to have access to the hint status of the extensions of server B? |
| 19:52.45 | tzafrir_laptop | justdave, do you refer to OSLEC? |
| 19:52.56 | justdave | hmm, maybe that's only for OSLEC, looks like there's a few others in there too |
| 19:52.56 | tzafrir_laptop | OSLEC was never in Zaptel/DAHDI |
| 19:53.55 | eppigy | justdave: we meet again |
| 19:54.31 | justdave | ok, looks like it did compile a few echo cancelers, I must have to list them in the MODULES thing in the config file to get them loaded or something? |
| 19:54.48 | justdave | and which one would be recommended? :) |
| 19:54.52 | [TK]D-Fender | justdave: this is the most boring, common and DOCUMENTED thing ever. |
| 19:55.09 | [TK]D-Fender | justdave: Just bloody-well install it like every other release of Zaptel. |
| 19:55.18 | justdave | well, it used to just work, and it stopped working when we switched from zaptel to dahdi |
| 19:55.36 | [TK]D-Fender | justdave: "Details would help. |
| 19:56.12 | justdave | details == I got lots of complaints from people about echoing on calls that come in via the TDM hardware since we upgraded |
| 19:57.15 | stabler | justdave: I believe he means conf's |
| 19:57.21 | justdave | if it's so well documented, I'd love to be pointed at it, because I've been up and down the documentation included in the tarball more times than I can count, and the OSLEC thing is the only mention of echo canceling at all in it. |
| 19:58.19 | [TK]D-Fender | justdave: Well I don't see you showing us your configs so we can see if you failed to do something properly. |
| 19:58.31 | [TK]D-Fender | justdave: And OSLEC is jsut another EC option |
| 20:00.28 | *** join/#asterisk jicksta (n=jicksta@c-67-169-165-162.hsd1.ca.comcast.net) |
| 20:00.47 | *** join/#asterisk voxter (n=voxter@76.77.95.2) |
| 20:02.21 | voipforces | Anyone? , I have 2 * server linked via IAX. anyway for server A to have access to the hint status of the extensions of server B? |
| 20:04.17 | [TK]D-Fender | voipforces: http://www.google.ca/search?hl=en&q=asterisk+distributed+presence&btnG=Google+Search&meta= |
| 20:04.56 | justdave | so which config would you like to see? without appropriate documentation it's hard to know what's relevant. :) |
| 20:05.09 | [TK]D-Fender | justdave: All of your DAHDI configs obviously. |
| 20:05.41 | [TK]D-Fender | justdave: Don't try and feed us some sort of cop-out. it IS documented in the source tarball like so much else... |
| 20:05.44 | [TK]D-Fender | :p |
| 20:06.04 | voipforces | [TK]D-Fender: great only in 1.6 again... |
| 20:06.35 | voipforces | [TK]D-Fender: based on 1.4 right now... |
| 20:08.07 | [TK]D-Fender | voipforces: You have your answer. If you are having trouble dealing with this revelation, #drphil awaits |
| 20:08.20 | voipforces | [TK]D-Fender: :-P |
| 20:08.22 | justdave | I'd be quite happy if you show me where. |
| 20:08.31 | justdave | FWIW there is a "doc" directory inside the tarball. |
| 20:08.32 | justdave | it's empty |
| 20:08.46 | justdave | perhaps that's the source of confusion. :) |
| 20:08.59 | justdave | (in the dahdi-linux-2.1.0.4.tar.gz) |
| 20:09.15 | [TK]D-Fender | justdave: /etc/dahdi/ , /etc/asterisk/ have files related to this. |
| 20:09.35 | justdave | but I did find some stuff in the config files while collecting them to show you that gave me a couple more places to look |
| 20:09.36 | voipforces | [TK]D-Fender: This could be an interesting hack: http://nuitari.org/asterisk/ |
| 20:12.42 | [TK]D-Fender | voipforces: Massively dirty hack witha monitoring script |
| 20:13.21 | voipforces | [TK]D-Fender: Yup, but might not have a choice for now. |
| 20:15.47 | justdave | ok, think I found enough to give me some things to try. |
| 20:15.58 | justdave | I'll be back later (with configs in hand next time) if I'm still stuck |
| 20:16.47 | *** join/#asterisk double_cheesburg (n=chatzill@ip70-162-238-155.ph.ph.cox.net) |
| 20:22.05 | *** join/#asterisk Neo2102 (n=Jarrings@91.176.37.179) |
| 20:22.08 | Neo2102 | Hello |
| 20:22.24 | Neo2102 | Who can help me ? All playback is played very slowwwwwwllly |
| 20:22.32 | Neo2102 | and it's not possible to hear somthing |
| 20:23.20 | voipforces | ztdummy problem I guess. |
| 20:23.44 | Neo2102 | I need to disable it ? |
| 20:23.55 | voipforces | which version of asterisk? |
| 20:23.59 | Neo2102 | 1.4 |
| 20:24.07 | voipforces | telephony hardware? |
| 20:24.18 | Neo2102 | 1.4.22 asterisk |
| 20:24.22 | Neo2102 | Siemens c470 |
| 20:24.36 | Neo2102 | but i've only this problem with my asterisk server on my lan |
| 20:24.43 | voipforces | I mean do you have a telephony card installed? |
| 20:24.45 | Neo2102 | when the outide asterisk I don't have problem |
| 20:24.57 | Neo2102 | No i'm sorry I've nothing installed |
| 20:25.14 | voipforces | Ok, you are not running asterisk in vmware I hope? |
| 20:25.18 | Neo2102 | yes |
| 20:25.30 | voipforces | you are running in vmware? |
| 20:25.31 | Neo2102 | I run asterisk on vmware:-) |
| 20:25.53 | voipforces | Ok, then make sure you have a USB hub configured in your vm achine |
| 20:26.37 | Neo2102 | It's vmware ACE |
| 20:27.01 | Neo2102 | I don't see usb option on vmware |
| 20:27.01 | voipforces | no mather, without a usb hub you will have timing issue |
| 20:27.09 | voipforces | usb hub that is. |
| 20:27.28 | Neo2102 | ok need to change vmware version with the lasted and enable usb hub ? |
| 20:27.32 | voipforces | when you edit your vm machine, you should be able to add a usb hub or device |
| 20:27.59 | voipforces | Never used vmare ace, I always use vmware server and never for production environement. |
| 20:28.11 | Neo2102 | it's only for testing |
| 20:28.20 | Neo2102 | I need to buy vmware Server or workstation ? |
| 20:28.48 | eppigy | Neo2102: esxi is free |
| 20:28.50 | voipforces | For testing and development I use the free VMWare Server (linux or windows) |
| 20:28.57 | eppigy | so is vmware server |
| 20:29.19 | jaytee | workstation costs money though |
| 20:29.46 | Neo2102 | vmware esxi is great ? |
| 20:30.04 | voipforces | I never used it. |
| 20:30.18 | Qwell | Neo2102: if I'm not mistaken, esxi is bare-metal |
| 20:30.33 | Qwell | but, vmware-server is awesome |
| 20:30.35 | Neo2102 | eppigy have you alreadu installed a asterisk on esxi ? |
| 20:30.37 | jaytee | never tried esxi myself, vmware server is great. I use it on this box for XP and on a Windows server for running sipX |
| 20:30.44 | Neo2102 | Okay |
| 20:30.46 | eppigy | Neo2102: why in the world would I do that |
| 20:30.57 | eppigy | I have test servers |
| 20:31.23 | Neo2102 | 1.0.8 I downlaod it and I can transfert the vmware hard drive or I need to reinstall all ? |
| 20:31.47 | eppigy | they have a vmware converter |
| 20:31.58 | eppigy | to virtualize physical servers |
| 20:32.10 | eppigy | if that is what youre trying to ask |
| 20:32.33 | voipforces | I thinks he means convertinf his ace images to server vmware images |
| 20:32.59 | voipforces | Neo2102: I would just re-install. A lot easier. |
| 20:33.14 | Neo2102 | yes it's really fast to reinstall |
| 20:33.18 | Neo2102 | thanks I will try that :-) |
| 20:33.32 | jaytee | unless you're running pure SIP I'd avoid running * on a VM and even if I was running pure SIP with no analog FXO or digital PRI cards in it I'd only run a VM if it was the "fastest hunk o junk in the galaxy" |
| 20:34.28 | voipforces | I would never use * in vmware for production period. |
| 20:34.55 | Neo2102 | my it's only for testing :-) I wait my dedicate server OVH |
| 20:35.34 | jaytee | voipforces, true dat! vm has it's place but in the world of voip in a production environment it's just begging for trouble |
| 20:36.54 | voipforces | jaytee: I use vmware for * for testing and development (code test) only. |
| 20:37.22 | *** join/#asterisk chrismaster1 (n=chrismas@chello080109200180.3.sc-graz.chello.at) |
| 20:44.45 | *** join/#asterisk madgeek (i=daemon@65-119-213-34.dia.static.qwest.net) |
| 20:46.39 | *** join/#asterisk chrismaster1 (n=chrismas@chello080109200180.3.sc-graz.chello.at) |
| 20:51.35 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 20:58.36 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 21:01.29 | *** join/#asterisk BadHAL (n=nn@cpe-72-179-194-139.stx.res.rr.com) |
| 21:02.56 | double_cheesburg | Hey I just installed asteriskNOW 64bit. Everything I've read says that zaptel.conf and zapata-auto.conf should be prebuilt with asterisk, but they aren't prebuilt with asteriskNOW. Does asterisk now still require these? |
| 21:04.44 | Qwell | double_cheesburg: 1.5.0? |
| 21:04.56 | double_cheesburg | Yeah, if that's the beta. Yeah |
| 21:05.17 | Qwell | it is, and it uses dahdi now |
| 21:05.27 | double_cheesburg | What is dahdi? |
| 21:05.38 | Qwell | replacement for Zaptel |
| 21:05.40 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 21:06.15 | double_cheesburg | chan_dahdi.conf.template |
| 21:06.17 | Qwell | double_cheesburg: basically, anything that references zaptel.conf now needs to be /etc/dahdi/system.conf, and zapata.conf becomes chan_dahdi.conf |
| 21:06.35 | double_cheesburg | so that's where I should be doing my zaptel.conf? |
| 21:06.43 | Qwell | no, /etc/dahdi/system.conf |
| 21:06.58 | Qwell | zaptel.conf was to configure zaptel itself, while zapata.conf was to configure Asterisk |
| 21:07.16 | double_cheesburg | is dahdi better? what's the difference? |
| 21:07.24 | double_cheesburg | I haven't read any documentation on this |
| 21:07.42 | Qwell | It's basically a copy of Zaptel, renamed, and split between drivers and utils, with a bunch of improvements |
| 21:08.29 | double_cheesburg | how about zapata.conf...same thing? |
| 21:08.46 | double_cheesburg | does dahdi take care of that too? |
| 21:09.04 | Qwell | that is now chan_dahdi.conf |
| 21:09.09 | Qwell | /etc/asterisk/chan_dahdi.conf |
| 21:09.22 | double_cheesburg | wow. is there a tutorial somewhere that talks about this? |
| 21:09.42 | double_cheesburg | I mean, I didn't see anything on the asterisk website |
| 21:09.44 | Qwell | I've never configured zaptel/dahdi with FreePBX.. |
| 21:09.52 | Qwell | for the hardware stuff, you should be able to get away with just running dahdi_genconf |
| 21:10.19 | Qwell | for Asterisk configs, you can use whatever instructions they have for zapata.conf (just keeping in mind the filename change) |
| 21:10.23 | mazpe | is there an industry standard or prefer extention for voicemail? |
| 21:10.38 | Qwell | mazpe: maybe standards. :) |
| 21:10.53 | mazpe | love it |
| 21:11.11 | mazpe | i guess make your own... and add it to the training manual |
| 21:11.13 | Qwell | 6000, 6060, 6245 (MAIL) |
| 21:11.19 | Qwell | mazpe: pretty much |
| 21:11.34 | mazpe | *98 freepbx |
| 21:11.38 | double_cheesburg | yeah, my dahdi directory contains 3 files. init.conf modules system.conf |
| 21:11.44 | Qwell | some systems let you dial your own # |
| 21:11.57 | Qwell | double_cheesburg: shouldn't need to touch the first 2 |
| 21:12.01 | mazpe | i guess i'm looking for something easy for the users as well |
| 21:13.25 | double_cheesburg | what a mess |
| 21:13.31 | Qwell | mazpe: what phones are you using? |
| 21:13.46 | Qwell | get Polycoms and let them just use the dedicated "Messages" button :p |
| 21:13.49 | mazpe | polycom 301 |
| 21:14.01 | Qwell | not sure if a 301 has them |
| 21:14.06 | mazpe | 301 doesnt have it |
| 21:14.11 | mazpe | low end phones |
| 21:14.35 | Qwell | double_cheesburg: if you want to make things easy, just use dahdi_genconf |
| 21:14.52 | *** join/#asterisk bijit (n=bijit@216.25.163.115) |
| 21:15.05 | double_cheesburg | how so? you mean create a file called dahdi_genconf ?? |
| 21:15.10 | double_cheesburg | and put what in it? |
| 21:15.12 | Qwell | no, it's a command |
| 21:15.20 | Qwell | run `man dahdi_genconf` |
| 21:15.42 | *** join/#asterisk SlicerDicer (n=SlicerDi@69-92-107-4.cpe.cableone.net) |
| 21:16.09 | bijit | how can I reboot all my aastra phones from asterisk? sip notify is for individual . |
| 21:16.10 | double_cheesburg | out of bash? |
| 21:16.13 | Qwell | double_cheesburg: yes |
| 21:16.30 | [TK]D-Fender | Qwell: My 301 does |
| 21:16.46 | Qwell | [TK]D-Fender: dunno, he says it doesn't |
| 21:16.48 | mazpe | light indicator |
| 21:17.11 | double_cheesburg | -bash: run: command not found |
| 21:17.12 | [TK]D-Fender | Qwell: go MAKE one. I did |
| 21:17.24 | mazpe | theres another option |
| 21:17.29 | *** join/#asterisk korihor (n=korihor@200.44.218.45) |
| 21:17.33 | mazpe | use one of the buttons |
| 21:17.50 | Qwell | [TK]D-Fender: tell him, not me :p I've never seen/used a 301, so no clue what it has |
| 21:18.42 | double_cheesburg | Qwell: I'm not sure what you mean. I simply get the previously stated error when running that command. |
| 21:18.52 | Qwell | double_cheesburg: man dahdi_genconf |
| 21:18.55 | Qwell | run that |
| 21:19.00 | *** part/#asterisk wacky_ (n=abourget@IP-208-88-110-46.mtl.fibrenoire.ca) |
| 21:19.17 | double_cheesburg | oh. the manual for dahdi_genconf. i see. sorry |
| 21:20.05 | double_cheesburg | Qwell: Yeah, I have no manual entry for dahdi_genconf |
| 21:20.42 | Qwell | yum install dahdi-tools-doc |
| 21:21.03 | double_cheesburg | sweet |
| 21:21.12 | Qwell | hopefully it's in there.. |
| 21:21.20 | *** join/#asterisk carpenike (n=ryan@c-98-218-125-247.hsd1.md.comcast.net) |
| 21:21.21 | double_cheesburg | yep |
| 21:21.55 | carpenike | Anybody know how to get app_rxfax on asterisk 1.4.23? |
| 21:23.32 | madsara | Is there any way I can force a SIP registration ? |
| 21:23.34 | *** join/#asterisk Greek-Boy (n=greek@41.222.89.77) |
| 21:23.53 | *** join/#asterisk harry__ (n=h@imperialglamour.com) |
| 21:23.57 | voipforces | carpenike: you need spandsp |
| 21:24.07 | harry__ | hey, do you know of any (cli) utility to convert mp3 -> alaw? |
| 21:24.21 | voipforces | carpenike: version 0.0.4pre16 |
| 21:24.24 | carpenike | voipforces: I have it installed. Don't I also need app_rxfax? |
| 21:24.47 | voipforces | yes it's in the spandsp site if I remember correctly. |
| 21:24.54 | harry__ | and using mp321/ffmpeg to convert mp3 -> wav then sox to do wav -> alaw seems stupid |
| 21:25.54 | harry__ | oh. I just answered my own question. ffmpeg can do mp3 -> alaw. sheesh |
| 21:27.19 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 21:27.54 | carpenike | voipforces: So I just installed that version of spandsp, when I go into make menuselect shoudl I see the app_fax? |
| 21:28.17 | voipforces | carpenike: Check ou this how-to: http://www.sems.org/entry.asp?ENTRY_ID=197 |
| 21:28.45 | voipforces | carpenike: Includes url to download app_rxfax |
| 21:29.50 | carpenike | hm, looks like the linkds are dead. |
| 21:30.10 | Greek-Boy | are priorities still carried out after Hangup() has occured? |
| 21:32.06 | manxpower | Greek-Boy: sure, h,1,whatever |
| 21:32.22 | *** join/#asterisk rpm (n=rpm@S010600055d2cf2e2.cg.shawcable.net) |
| 21:33.12 | *** join/#asterisk Neo2102 (n=Jarrings@91.176.37.179) |
| 21:34.28 | *** join/#asterisk supjigator (n=shanebur@64.147.216.10) |
| 21:37.01 | rpm | handle_response_invite: Failed to authenticate on INVITE to '"asterisk" <sip:asterisk@x.x.x.x> whats this actually mean? |
| 21:37.54 | manxpower | rpm: usually that means the device connected as user "asterisk" and you don't have an [asterisk] section of SIP.conf |
| 21:40.44 | rpm | in my register => line, i am specifying 'jive' for the extension, shouldn't it be looking there? |
| 21:40.53 | *** join/#asterisk bijit (n=bijit@216.25.163.115) |
| 21:42.45 | *** join/#asterisk Firass-z0r (n=asadf@c-67-201-205-34.reshall.wwu.edu) |
| 21:43.15 | Greek-Boy | thanks maxpower |
| 21:43.36 | Greek-Boy | when using a Gotoif what is the difference between '&' and "&&" ? |
| 21:43.42 | Greek-Boy | is & like "or" |
| 21:43.50 | Greek-Boy | and is && like "and" ? |
| 21:44.42 | Neo2102 | voipforces now Asterisk is installed |
| 21:44.46 | Neo2102 | on Vm Server |
| 21:44.48 | Neo2102 | problem is not resolved |
| 21:44.54 | Neo2102 | I've added a USB Card |
| 21:45.09 | Neo2102 | Hub |
| 21:45.21 | voipforces | Neo2102: hmm. can not help you right now have to go on a site. |
| 21:45.38 | Neo2102 | Any other test possible to see what's wrong ? |
| 21:45.48 | Neo2102 | I've just installed Elastix |
| 21:45.58 | Neo2102 | I've configure nothing yet inly 1 extension. |
| 21:46.37 | voipforces | Neo2102: Is your usb in auto-connect? if not put it on. Also Try to mount a USB key on your vmware session, sometimes it helps. |
| 21:46.42 | voipforces | Have to go now. |
| 21:47.01 | Neo2102 | Ty |
| 21:49.21 | rpm | anyone here use jive communications with asterisk? |
| 21:49.25 | *** part/#asterisk harry__ (n=h@imperialglamour.com) |
| 21:51.35 | Neo2102 | No sorry |
| 21:51.47 | Neo2102 | rpm do you have idea all playback sound is slow |
| 21:52.18 | rpm | nope. |
| 22:03.04 | *** join/#asterisk Ritzerisk (n=Ritztech@nv-65-40-156-46.sta.embarqhsd.net) |
| 22:03.17 | Ritzerisk | i understand the the asterisk is a sip proxy |
| 22:04.47 | eppigy | 8[] |
| 22:05.04 | Ritzerisk | is there a way to just use it as a sip proxy and nothing else |
| 22:05.55 | manxpower | slow playback is frequently caused by ztdummy and a bad kernel HZ |
| 22:06.09 | manxpower | ztdummy wants the kernel HZ to be 1000 |
| 22:06.25 | manxpower | Ritzerisk: Asterisk is NOT a SIP Proxy |
| 22:06.35 | Neo2102 | manxpower I've installed Elastix |
| 22:06.47 | Neo2102 | On vm-ware |
| 22:06.57 | manxpower | Neo2102: I can't help you with that. I can only help you with standard Asterisk. I also cannot help you with VMware + Asterisk |
| 22:07.01 | Ritzerisk | soo how does it translate the private address in the sdp message to the external address |
| 22:07.06 | manxpower | So, we have reached the extent of my support |
| 22:07.17 | Neo2102 | ztdummy I need to disabled it ? |
| 22:07.22 | manxpower | Ritzerisk: that's not a proxy thing. Asterisk is a B2BUA |
| 22:07.32 | manxpower | Neo2102: do you need IAX2 trunking or MeetMe? |
| 22:07.41 | Neo2102 | No |
| 22:08.06 | Neo2102 | how I can disable this ?N |
| 22:08.11 | manxpower | Then you don't need a Zaptel / ztdummy if you don't need those two features. |
| 22:08.22 | manxpower | You edit /etc/sysconfig/zaptel and comment it out there. |
| 22:11.21 | rpm | does asterisk not know how to respond to SIP 407's? |
| 22:11.45 | rpm | It gets the challenge, but does notthing |
| 22:12.19 | *** join/#asterisk path_ (n=path@pc-15-190-86-200.cm.vtr.net) |
| 22:13.33 | eric256 | googling "asterisk sip 407" results in quite a few results most of which seem to have answers. |
| 22:14.55 | *** join/#asterisk verywiseman (n=verywise@unaffiliated/verywiseman) |
| 22:16.48 | eric256 | Anyone used a Cisco CP-7910G? |
| 22:17.48 | Greek-Boy | eric256 I did a while back |
| 22:17.49 | Greek-Boy | why? |
| 22:18.09 | eric256 | just curious if it was any good? found them for 40$ and wondered |
| 22:18.45 | manxpower | eric256: does it already have the SIP firmware on it? |
| 22:19.23 | eric256 | doesn't specify http://www.voipsupply.com/cisco-cp-7910g |
| 22:19.51 | manxpower | eric256: if it comes with the phone, power supply, and SIP firmware then $40 is a good price. |
| 22:19.54 | manxpower | If not then it's not. |
| 22:20.29 | eric256 | thanks i might have to call and see. the website is a bit lacking in those details |
| 22:23.08 | bmoraca | the power supplies usually sell for $10-$15 each...that said, the 7910 sucks |
| 22:25.17 | eric256 | hehe yea i thought it might for that price. whats the cheapest phone you'd go with? (without being overly cheap) |
| 22:25.28 | bmoraca | Polycom 330 |
| 22:25.33 | eric256 | its hard to tell with all these different models whats good and whats overpriced ;) |
| 22:25.42 | bmoraca | or Cisco 7912G...but probably the Polycom 330 |
| 22:26.18 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 22:27.09 | eric256 | these things are still pretty cheap after working with Vodavi or Intertel, where the phone was like 300$ and then a phone license and a server license and blah blah blah ;) |
| 22:27.51 | *** join/#asterisk riddlebox (n=user@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:28.23 | bmoraca | http://www.voiplink.com/POLYCOM_330_p/polycom-330.htm |
| 22:29.26 | *** join/#asterisk BuSyAnToS (n=31749@81-208-83-253.fastres.net) |
| 22:29.34 | bmoraca | or http://www.voiplink.com/POLYCOM_320_IP320_p/polycom-320.htm if you don't need the ethernet passthrough |
| 22:30.26 | *** join/#asterisk ingenius (n=alektro@host251.190-31-44.telecom.net.ar) |
| 22:30.49 | eric256 | thanks |
| 22:31.14 | verywiseman | is asterisk has "Automatic Hold" feature? |
| 22:31.42 | *** join/#asterisk edwin_quijada (n=macaruch@200.26.172.98) |
| 22:31.43 | cjk | hi, i seem to have a problem where the fax machine connected to an fxs port sends the number but it arrives differently (few digits are wrong) in asterisk, has anyone seen this? |
| 22:32.14 | edwin_quijada | Hi! Somebody hear has used NVLineDetect for Asterisk 1.4 |
| 22:32.33 | edwin_quijada | I am making an autodialer |
| 22:33.06 | edwin_quijada | and I need to know the line status |
| 22:33.58 | EmleyMoor | cjk: Is it attempting to pulse dial? |
| 22:34.14 | eric256 | wow they have some 802.11g phones that look just like cell phones. lmao, thats realy pretty cool |
| 22:34.55 | edwin_quijada | eric256: where i can find it ? |
| 22:35.38 | *** join/#asterisk Chuggs (n=Chuggs@s142-179-186-158.ab.hsia.telus.net) |
| 22:35.47 | eric256 | wireless phones? http://www.voiplink.com/Wireless_IP_Phones_s/37.htm |
| 22:36.33 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 22:37.33 | bmoraca | eric256: would recommend staying away from those. |
| 22:37.41 | bmoraca | And the Snom M3 as well |
| 22:37.48 | *** join/#asterisk _pepo_ (n=__pepo__@200.55.224.2) |
| 22:38.01 | bmoraca | if you need wireless, get a Linksys PAP2T or other media gateway and use a standard analog DECT phone |
| 22:38.04 | *** join/#asterisk tamiel (n=tamiel@ip-176.net-89-2-117.rev.numericable.fr) |
| 22:38.07 | _pepo_ | hi friends |
| 22:38.39 | eric256 | what about grandstream or astra? |
| 22:39.06 | bmoraca | grandstream and aastra are cheap phones. some like them. i personally find them cheap. i stick to polycom and cisco |
| 22:39.13 | *** join/#asterisk murdock_ut (n=chatzill@mail.kimballequipment.com) |
| 22:39.44 | eric256 | sometimes cheap is the right fit ;) |
| 22:39.48 | EmleyMoor | I am seriouly thinking Linksys |
| 22:39.55 | EmleyMoor | seriously |
| 22:39.57 | bmoraca | Linksys phones are REALLY cheap |
| 22:40.15 | bmoraca | i didn't like them at all |
| 22:40.27 | Greek-Boy | can one use pattern matching for the variable ${CALLERID(num)} I wanna use it in a Gotoif... |
| 22:40.54 | carrar | LinkSys phones have a lot more cofiguration capability then the Cisco phones |
| 22:41.01 | carrar | te SPA94X |
| 22:41.14 | bmoraca | carrar: that may be, but they're really poor quality |
| 22:41.23 | carrar | They work great |
| 22:41.36 | carrar | I've not had issues with them with out customers |
| 22:41.37 | EmleyMoor | Polycom would be my other choice |
| 22:41.37 | bmoraca | carrar: i didn't say they didn't work. i said they were poor quality. |
| 22:41.50 | carrar | in what way |
| 22:41.54 | voxter | anyone know how many packets per second an IAX2 call takes? |
| 22:41.57 | voxter | using g729a |
| 22:42.31 | bmoraca | carrar: the speakerphone sounds like two tuna fish cans stuck together with a string. and the handset isn't that much better. |
| 22:42.42 | Neo2102 | Hep |
| 22:42.53 | _pepo_ | please, with ${EXTEN} I have the extension but: How do I obtain the field TO: of the SIP header? |
| 22:42.57 | Neo2102 | Who can told me why when I hear playback it's toooooooo slowly !!! |
| 22:43.19 | bmoraca | Neo2102: bad timing source? |
| 22:43.20 | carrar | but they aren't cheap quality |
| 22:43.26 | EmleyMoor | Neo2102: What are you playing back? |
| 22:43.31 | carrar | sound isn't as good as polycom/cisco |
| 22:43.35 | carrar | nor the feel |
| 22:43.42 | Neo2102 | the playback it's the default sound |
| 22:43.43 | carrar | thats a given |
| 22:43.48 | Neo2102 | when I dial *97 |
| 22:43.51 | Neo2102 | or somthing like that |
| 22:44.05 | eric256 | Neo2102: timing on VMMachines is notoriously bad |
| 22:44.21 | Neo2102 | it's because I run it on vm ? |
| 22:44.29 | manxpower | Neo2102: stop asterisk. run rmmod ztdummy then start Asterisk |
| 22:44.33 | eric256 | i've got some that will loose like 20 hours every 24 hours if they arn't configured right |
| 22:44.43 | manxpower | if that doesn't work than that's not the problem |
| 22:44.52 | rpm | jesus that was dirty, i had to put in 4 different auth= statements to get this sip trunk to work |
| 22:45.32 | eric256 | i dunno if that is your problem, but i wouldn't run anything time or speed sensitive in a vm machine. if you can it always helps to turn of variable CPU speed on the host. (normaly in the energy settings somewhere) |
| 22:45.43 | jameswf | <troll> Qwell <~~http://trixbox.org/forums/trixbox-ce-development-forum/beta-forum/asterisk-16-0 </troll> |
| 22:45.49 | bmoraca | i've got asterisk running on top of ESXi and i lose about a quarter second per day |
| 22:45.50 | eric256 | off* |
| 22:46.04 | eric256 | yea ESXi is a whole different beast ;) |
| 22:46.06 | Greek-Boy | _pepo_ aren't u looking for ${SIPCHANINFO(uri)} ? |
| 22:46.16 | eric256 | VMWare Server can be a dog |
| 22:46.48 | eric256 | i have 3 ESXi too and they are awsome and even some VMWare servers running great. |
| 22:48.02 | *** join/#asterisk [TK]D-Fender (n=zsirc@161.216.191.60) |
| 22:48.15 | _pepo_ | nop ${SIPCHANINFO(uri)} give me the FROM: field |
| 22:48.32 | _pepo_ | i need TO: field |
| 22:49.43 | Greek-Boy | try ${SIP_HEADER(TO) |
| 22:49.58 | Greek-Boy | i mean ${SIP_HEADER(TO)} |
| 22:51.07 | Greek-Boy | _pepo_: any luck? |
| 22:51.18 | _pepo_ | :( |
| 22:52.51 | _pepo_ | ${SIP_HEADER(TO)} has all the fiel... now I have to filter... thanks |
| 22:53.20 | Greek-Boy | great stuff |
| 22:53.31 | Greek-Boy | at least you're getting closer :-) |
| 22:53.55 | _pepo_ | ;) |
| 22:55.24 | carrar | Why do you want to pull it out of headers? |
| 22:56.03 | double_cheesburg | does anyone know of a good tutorial on how to configure dahdi? |
| 22:56.25 | rene- | dahdi generates its own configuration |
| 22:56.29 | [TK]D-Fender | carrar: he's too l33t for ${exten} :) |
| 22:56.30 | *** join/#asterisk qdk (n=qdk@0x573d8dd1.bynqu2.dynamic.dsl.tele.dk) |
| 22:56.37 | carrar | hahah tk |
| 22:56.45 | [TK]D-Fender | ~dahdi |
| 22:56.46 | jbot | [~dahdi] Digium/Asterisk Hardware Device Interface (DAhdi). The new name of zaptel More info at http://www.asterisk.org/zaptel-to-dahdi , and is pronounced "dah-dee" with a short A, or pronounced like http://www.russellbryant.net/dahdi.wav |
| 22:57.25 | rene- | how hard would be to create an ethernet switch from scratch |
| 22:57.34 | [TK]D-Fender | double_cheesburg : and in the tarball |
| 22:57.39 | rene- | or an ethernet interface |
| 22:57.54 | carrar | build a NIC Card? |
| 22:58.01 | rene- | yes |
| 22:58.02 | rene- | or switch |
| 22:58.07 | bmoraca | REALLY hard |
| 22:58.13 | bmoraca | why on earth would you want to do that? |
| 22:58.14 | carrar | if you got the parts |
| 22:58.28 | rene- | well, i d like to build poe switches |
| 22:58.32 | bmoraca | a hub would be relatively easy...a switch would be much more difficult |
| 22:58.35 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:58.39 | carrar | rene- thats not too hard |
| 22:58.45 | carrar | just use a existing switch |
| 22:58.52 | carrar | and add the PoE |
| 22:58.58 | rene- | and profit?? |
| 22:58.59 | carrar | modify the sockets |
| 22:59.07 | carrar | easier to buy one however |
| 22:59.15 | [TK]D-Fender | rene-: "to bake an apple pie from scratch one must first create the Universe" |
| 22:59.34 | bmoraca | lol |
| 22:59.36 | bmoraca | there you go |
| 22:59.47 | rene- | what i want to do,is be able to stick a power ethernet switch into a computer case |
| 22:59.49 | carrar | but most PoE switches give you logical control of the PoE status and power draw |
| 22:59.55 | carrar | which is nice |
| 23:00.11 | [TK]D-Fender | s/nice/necessary |
| 23:00.18 | rene- | like mitel does |
| 23:00.23 | rene- | or cisco etc |
| 23:00.34 | EmleyMoor | I am buying a PoE switch one day soon - in preparation for getting more phones |
| 23:00.37 | carrar | just go buy a nice Cisco switch off ebay that PoE |
| 23:01.09 | bmoraca | if you don't need to power Cisco phones or don't need full 15.4W over all ports, even a used Cisco PoE is more expensive than some of the stuff you can get from other vendors |
| 23:01.15 | carrar | then you can do QoS also if you have more then just phones plugged in |
| 23:01.25 | *** part/#asterisk eric256 (n=ejh@72.1.122.62) |
| 23:01.29 | [TK]D-Fender | carrar: many cisco's are proprietary PoE |
| 23:01.36 | carrar | the older ones |
| 23:01.40 | carrar | not the newer ones |
| 23:01.51 | rene- | i like how the openwrt has a switch and splits it into vlans as if they were independent interfaces |
| 23:01.58 | EmleyMoor | NetGear do some small PoE switches - and I intend to use standard PoE phones |
| 23:02.26 | [TK]D-Fender | carrar: the rest are just ridiculously overpriced :) |
| 23:02.39 | carrar | yeah, but they are nice |
| 23:02.41 | bmoraca | EmleyMoor: just be careful. NetGear severely limits the amount of power it delivers via PoE |
| 23:02.54 | rene- | btw, i am consistently blowing up dlink 1228's |
| 23:02.58 | rene- | with poe |
| 23:02.58 | bmoraca | a 3548 PoE does standard 802.11af and isn't all THAT expensive |
| 23:03.02 | [TK]D-Fender | I find d-link |
| 23:03.20 | [TK]D-Fender | and linksys poe swithes Ok |
| 23:03.30 | rene- | dlink said that they were getting bricked because of me using all the ports as POE |
| 23:03.44 | rene- | they just go all red lights |
| 23:03.51 | rene- | dlink is replacing them |
| 23:03.53 | carrar | haha |
| 23:03.54 | rene- | i bought over 20 |
| 23:03.57 | [TK]D-Fender | bmoraca:how much? |
| 23:04.00 | bmoraca | my switch of choice for cheap PoE is usually Dell. they suck for any kind of managed application, but they do full 15.4 over all ports and are pretty decent |
| 23:04.03 | carrar | nice "We support PoE, just not on all ports" |
| 23:04.07 | EmleyMoor | bmoraca: I will of course check for alternatives |
| 23:04.31 | rene- | bmoraca: i will be switching to dell, |
| 23:04.51 | rene- | but i have had some of the cisco cheaper end stuff failing on me |
| 23:04.54 | bmoraca | Cisco 3524 PoE for $149: http://cgi.ebay.com/CISCO-WS-C3524-PWR-XL-EN-24-PORT-3500-POE-POWER-SWITCH_W0QQitemZ220359662147QQcmdZViewItemQQptZCOMP_EN_Routers?hash=item220359662147&_trksid=p3286.c0.m14&_trkparms=72%3A1234%7C66%3A2%7C65%3A12%7C39%3A1%7C240%3A1318%7C301%3A1%7C293%3A1%7C294%3A50 |
| 23:05.00 | rene- | the catalyst express series |
| 23:05.15 | bmoraca | catalyst express has lifetime warranty |
| 23:05.32 | rene- | very nice |
| 23:05.39 | carrar | We have a few customers with WS-C3560-48PS |
| 23:05.40 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 23:05.41 | rene- | really unexpensive |
| 23:05.43 | carrar | they work nice |
| 23:05.49 | bmoraca | carrar: no doubt |
| 23:06.03 | bmoraca | carrar: i like the 3570s because they can do cross-chassis EtherChannel |
| 23:06.08 | carrar | yeah |
| 23:06.11 | carrar | $$$ |
| 23:06.14 | carrar | $$$++ |
| 23:06.17 | carrar | heh |
| 23:06.19 | bmoraca | lol |
| 23:06.20 | rene- | i found one day some israeli poe injectors, the price was really cheap 24 port for like 50USD, shipment was costlier than the device |
| 23:06.23 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 23:06.23 | carrar | 37 anything is nice |
| 23:06.31 | carrar | oh you said 35 |
| 23:06.40 | manxpower | rene-: just make sure they are not going to "bomb out" 8-| |
| 23:06.43 | manxpower | ducks |
| 23:06.48 | bmoraca | yeah...3570 switch. still really expensive |
| 23:06.56 | bmoraca | erm |
| 23:06.57 | bmoraca | god |
| 23:06.59 | carrar | TK, WS-C3560-48PS on ebay is around $2k |
| 23:07.00 | bmoraca | lysdexia |
| 23:07.04 | bmoraca | it is 3750 |
| 23:07.18 | [TK]D-Fender | carrar:obscene |
| 23:07.31 | carrar | 48 ports of PoE goodness |
| 23:07.48 | EmleyMoor | I'm looking for switches costing tens, not hundreds, of GBP |
| 23:07.49 | carrar | with QoS |
| 23:07.55 | [TK]D-Fender | carrar:d-links run me around$400 @24 port |
| 23:07.56 | bmoraca | carrar: Cisco 3560G 48port Gigabit PoE switch...https://shop.horizondatacom.com/oasiscrm/Shopping.asp?Directive=InvDetails&Parent=&Category=Switches&Selection=000178 |
| 23:08.02 | bmoraca | $4100 |
| 23:08.10 | carrar | You don't get the management you get with Cisco |
| 23:08.15 | EmleyMoor | I only intend to run two phones |
| 23:08.22 | EmleyMoor | (in here) |
| 23:08.37 | bmoraca | EmleyMoor: what router do you have? Cisco ASA5505's have 2 PoE ports built in to them |
| 23:08.44 | [TK]D-Fender | EmleyMoor:you clearly need TWO then :p |
| 23:08.57 | jameswf | write a check... |
| 23:09.14 | EmleyMoor | Some heap of crap by Thomson |
| 23:09.23 | rene- | TK: do u use all your ports on your 1228's? |
| 23:09.30 | rene- | and have they bricked on u? |
| 23:09.39 | [TK]D-Fender | rene-: I |
| 23:10.03 | [TK]D-Fender | 'm on the older 1526's |
| 23:10.05 | EmleyMoor | If I get a new house I might arrange for something better |
| 23:10.39 | [TK]D-Fender | rene-about 2/3 full x2 |
| 23:11.00 | [TK]D-Fender | brb |
| 23:11.08 | bmoraca | EmleyMoor: why not get one-port injectors then? you can usually pick them up for cheap...$30-40 each |
| 23:11.26 | rene- | hmm i have them all full |
| 23:11.52 | EmleyMoor | bmoraca: Wasteful of power points |
| 23:12.08 | obnauticus | [TK]D-Fender that's a nice switch |
| 23:12.09 | bmoraca | i agree, lol, but at 2 devices, it doesn't really matter |
| 23:12.22 | obnauticus | prestandard PoE |
| 23:12.23 | obnauticus | ew |
| 23:12.25 | *** join/#asterisk path_ (n=path@pc-15-190-86-200.cm.vtr.net) |
| 23:12.32 | carrar | rene, sounds like you need to make your own switch from scratch!! |
| 23:12.39 | bmoraca | lol |
| 23:12.40 | obnauticus | lmao |
| 23:12.42 | rene- | hehe |
| 23:12.43 | obnauticus | Hey |
| 23:12.46 | obnauticus | tony strak built himself in a cave |
| 23:13.25 | rene- | i am like years away from devising such tech hehe |
| 23:13.34 | obnauticus | it's just multiplexing |
| 23:13.38 | obnauticus | and some backend software |
| 23:13.41 | obnauticus | you could do it with an FPGA |
| 23:13.48 | carrar | heh |
| 23:13.53 | carrar | go buy a videocard |
| 23:13.55 | carrar | and mod it |
| 23:13.55 | rene- | yes FPGA |
| 23:13.57 | obnauticus | how the pros do it |
| 23:14.07 | bmoraca | i had a dell 24pt PoE switch go bad...ordered the advanced replacement when i bought it...and dell didn't want to replace it...yelled at them and they sent me out a 48 port |
| 23:14.20 | obnauticus | carrar are video cards modble into Digital IO |
| 23:14.25 | EmleyMoor | bmoraca: I would consider that answer if it weren't for the proximity and the lack of power outlets |
| 23:14.31 | carrar | SURE! |
| 23:14.34 | carrar | no idea |
| 23:14.37 | obnauticus | facepalm. |
| 23:14.50 | *** join/#asterisk zchaos (n=none@CPE001d7ef0ba9d-CM001ceab63f9a.cpe.net.cable.rogers.com) |
| 23:14.51 | carrar | it's all 's and 0's |
| 23:14.54 | carrar | 1's |
| 23:14.58 | EmleyMoor | In a new house I would want to allow for more phones anyway |
| 23:15.02 | carrar | mostly 0's |
| 23:15.35 | bmoraca | carrar: switching implies logic, though. a hub (or mux) is purely electrical |
| 23:15.47 | obnauticus | .... |
| 23:16.05 | carrar | ok, but we're talking about building a switch :) |
| 23:16.23 | obnauticus | You could do it with some MUX'ing equptment and an FPGA |
| 23:16.29 | obnauticus | but you have to really know what you are doing |
| 23:16.29 | carrar | HUB's are so 80's! |
| 23:16.32 | obnauticus | to get it 100mbit |
| 23:16.33 | obnauticus | lol |
| 23:16.56 | bmoraca | why would you want to reinvent the wheel? Cisco's got good switches already. |
| 23:17.15 | rene- | i just want cheap device to bolt into the side of my small asterisk box |
| 23:17.20 | Qwell | bmoraca: depends on your definition of good |
| 23:17.33 | rene- | 4 ports or 8 ports |
| 23:17.42 | carrar | Use large bolts |
| 23:17.51 | carrar | small ones won't work |
| 23:17.54 | EmleyMoor | When I move I am getting the new place wired up |
| 23:17.55 | bmoraca | ok, so you don't like Cisco...buy a foundry or extreme instead |
| 23:18.26 | carrar | You cna find floundary switches for free sometimes |
| 23:18.34 | obnauticus | where? |
| 23:18.40 | obnauticus | I have a BigIron i don't use. |
| 23:18.40 | obnauticus | lol |
| 23:18.52 | obnauticus | a MG8 |
| 23:18.56 | carrar | people get ride of their old meltware |
| 23:19.08 | EmleyMoor | That's a Marty Robbins song isn't it? <g> |
| 23:20.39 | Khratos | going home |
| 23:20.46 | *** part/#asterisk Khratos (n=khratos@190.166.103.227) |
| 23:22.09 | rpm | does asterisk still look for sounds in /var/lib/asterisks/sounds ? |
| 23:22.12 | *** join/#asterisk path_ (n=path@pc-15-190-86-200.cm.vtr.net) |
| 23:22.14 | Qwell | rpm: yes |
| 23:22.23 | Qwell | I think by default, 1.6 adds en/ |
| 23:22.32 | Qwell | (then falls back if they aren't there) |
| 23:22.56 | carrar | Whats wrong with a bazillon sound files in one directory!! :) |
| 23:23.11 | EmleyMoor | I have the Jay Benham sounds on mine (as well as Allison) |
| 23:23.48 | *** part/#asterisk Deeewayne (n=dwayne@nat/digium/x-90f9beab77754284) |
| 23:26.03 | rpm | hmm.. it's not finding any of my sound-bytes |
| 23:34.49 | *** join/#asterisk harry_v (n=lork@S010600a0c93f6f7e.vs.shawcable.net) |
| 23:36.49 | *** join/#asterisk jayrod422 (n=jayrod42@pool-71-245-188-175.pitbpa.fios.verizon.net) |
| 23:38.15 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 23:42.07 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 23:42.54 | beek | Evening jaytee and [TK]D-Fender |
| 23:48.07 | jaytee | evening beek |
| 23:48.25 | beek | How's it going? |
| 23:49.01 | jaytee | pretty good! I got my Big Lebowski "The Dude" action figure in the mail today so I'm psyched. :-) |
| 23:49.33 | *** join/#asterisk cheGGo (n=cheGGo2@dslb-088-069-027-222.pools.arcor-ip.net) |
| 23:49.38 | *** join/#asterisk path_ (n=path@pc-15-190-86-200.cm.vtr.net) |
| 23:49.50 | beek | googles |
| 23:50.43 | beek | Ahhh... perhaps I'm living in the backwaters of the country but I'd never heard of him. |
| 23:51.02 | jaytee | http://www.rock-n-roll-action-figures.com/20sdexbiglea.html |
| 23:51.30 | beek | You a collector? |
| 23:51.41 | jaytee | no, I'm just a fan of the movie |
| 23:52.21 | beek | I'll have to go see that. |
| 23:53.08 | jaytee | it's been out on DVD for almost a decade now |
| 23:53.28 | beek | Add it to the netflix queue... |
| 23:54.25 | beek | Phone system got a good workout last week. We had severe windstorms here and handled 3500 calls in 24 hours. Not huge, by any stretch, but a good test for it. No PRI problems and no complaints. |
| 23:54.25 | jaytee | and there are all kinds of clips of it on YouTube. it's kinda got a cult following. Jeff Bridges is "The Dude", John Goodman plays his friend Walter who has serious anger management issues and Steve Buscemi plays their bowling partner, Donny. |
| 23:54.56 | beek | Maybe I can seek out all of the clips and assemble the complete movie? ;-) |
| 23:55.09 | jaytee | I'm now at 151 sip peers (phones) and will be topping out at 174 at the end of next week and pulling the plug on my Option 11C |
| 23:55.23 | beek | Get all of your lines toned out? |
| 23:55.31 | harry_v | Hi jaytee :) |
| 23:55.33 | jaytee | be advised the movie is definitely not for children |
| 23:55.40 | jaytee | hi harry_v |
| 23:55.45 | beek | Not a problem... |
| 23:56.01 | *** join/#asterisk Failrar (n=Failrar@fsm.xs4all.nl) |
| 23:56.21 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 23:56.23 | jameswf | jaytee: you could be an action figure http://www.vicale.com/maleondemand/ |
| 23:56.24 | jaytee | there's one clip on YouTube where they took all the swearing parts and mashed em together. it's 8 minutes long :-) |
| 23:56.32 | beek | jaytee: How did you set that up? Separate ports for the phones? |
| 23:57.12 | harry_v | I wonder how many creative ideas ast has been used for. RFID call notificatetion to front nurse desk if baby taken. That would be interesting. Anyone seen awhere a db driven web site is opened automatily with cid account number showing? |
| 23:57.42 | harry_v | What movie? |
| 23:58.15 | jaytee | beek, pardon? I'm using 82 Polycoms, 76 are 330's, 6 are 550's and the remainder of the phones are analog connected by either a Linksys SPA2102 or an SPA8000. |
| 23:58.35 | harry_v | jaytee, you have a auto provison script? |
| 23:58.40 | jaytee | harry_v, The Big Lebowski |
| 23:58.57 | jaytee | harry_v, I have two scripts, one for 550's and one for 330's. |
| 23:59.04 | harry_v | I wouldnt mind going to a film festival |
| 23:59.21 | beek | I was wondering if you have separate ports for each phone at the desk, with separate switches, or if you're sharing the data/VoIP on a single port. |
