| 00:00.43 | Powerkill | when i do rtp debug i see only one IP |
| 00:01.28 | drmessano | Why do you have two different public IPs for SIP and RTP? |
| 00:02.14 | [TK]D-Fender | shamess, Sure, Wordpress is just MySQL. I'm sure you could dummy something up to take your recording, move it in a likable place and issue a "post" |
| 00:02.57 | shamess | Yeah, I can make Wordpress do what I want. I just don't know how to go about hte "phoning my server" bit. |
| 00:05.15 | [TK]D-Fender | shamess, what would the call come in on? |
| 00:05.38 | *** join/#asterisk PepOSX (n=angeldav@200.93.28.168) |
| 00:06.37 | shamess | What ever I could make it work on, really. What would be ideal would be some sort of Skype client that I could use to call it. |
| 00:07.13 | [TK]D-Fender | shamess, Skype isn't really viable..... but any other number of means would. |
| 00:07.26 | JayTee52 | shamess, am i understanding correctly, you want to use voice to input to a blog? |
| 00:07.48 | [TK]D-Fender | shamess, Direct SIP/IAX calls, ITSP, land-line, etc |
| 00:08.44 | shamess | Yeah, I didn't mean Skype specifically, just an application sort of like that. Oh. Umm I'm not sure what most of those acronyms mean. I guess land-line wouldn't work, because my server would really have a phone number, ne? |
| 00:09.25 | [TK]D-Fender | shamess, Sure it would work. the question is how do you want to have these "calls" come in to have notes recorded? |
| 00:09.39 | shamess | JayTee52: Yeah. I want to be able to phone something, have that record what I've said as a sound file, so I can have it moved to a web directory. |
| 00:10.08 | JayTee52 | oh, so the blog would be a link to a wav file or other audio format? |
| 00:10.16 | [TK]D-Fender | shamess, if its just you you could set up a soft-phone and record right from your PC. |
| 00:10.19 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 00:10.21 | [TK]D-Fender | ~softphone |
| 00:10.21 | jbot | [~softphone] A soft-phone is a program that lets your computer use a compatible audio interface (usually a speaker&mic, or headset), and optionally a webcam, to place VoIP calls with a given protocol (SIP/IAX/H.323/MGCP/etc). For links enter ~zoiper , ~xlite , ~twinkle , ~bria , ~eyebeam |
| 00:11.22 | shamess | Oh, I think that's just what I want [TK]D-Fender :) |
| 00:11.34 | JayTee52 | I've used X-Lite and messed around a little with Zoiper but I've always avoided twinkle because I was afraid the other guys would make fun of me. :-) |
| 00:11.43 | shamess | hehe |
| 00:11.45 | [TK]D-Fender | shamess, then go download *, a soft-phone, and the BOOK, and get busy |
| 00:11.47 | [TK]D-Fender | ~book |
| 00:11.48 | 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, or see ~buybook |
| 00:12.31 | shamess | Cool, I'll go read up then. Thanks for the help, guys :) |
| 00:13.54 | *** join/#asterisk RypPn_OuT (i=TuMbL@rosscom.demon.co.uk) |
| 00:14.38 | shamess | Just so I'm not going off on the wrong direction; I'd download the softphone as a client (for my Windows machine) and put * on my server? |
| 00:14.51 | [TK]D-Fender | shamess, yup |
| 00:16.20 | plik | shamess: you could also have a phone number point to your server so yu could dial-a-post from any phone |
| 00:16.47 | shamess | That would be incredibly awesome, plik. But I imagine that to be expensive, ne? |
| 00:16.59 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 00:17.22 | plik | depends where you are - some place do free DID numbers, some cahrge a few $ per month |
| 00:17.28 | drmessano | Get unlimited incoming from les.net for $4 a month |
| 00:17.36 | [TK]D-Fender | shamess, Depends how you take in your calls. You could get a Washington state # for free over SIP via www.ipkall.com |
| 00:17.37 | drmessano | or get an IPKALL number |
| 00:17.55 | plik | IPKALL thats the free washington no isn't it? |
| 00:18.00 | drmessano | yeah |
| 00:18.08 | *** join/#asterisk dacs (n=haiger@unaffiliated/dacs) |
| 00:18.13 | JayTee52 | IPKALL......echocancellation=no |
| 00:18.28 | shamess | Well, I'm in England, but my server is in America, so I'm not sure how it would work best. |
| 00:18.36 | [TK]D-Fender | JayTee52, Don't like their service? Then they'll give you double your money back! |
| 00:18.54 | [TK]D-Fender | shamess, Where would the calls be coming from? |
| 00:19.04 | shamess | England. |
| 00:19.09 | plik | shamess: voiptalk.org will do you a uk landline number for £1.99 per monh |
| 00:19.20 | JayTee52 | [TK]D-Fender, I was just making a bad * pun because you and drmessano both said IPKALL almost at the same time, like echo. |
| 00:19.25 | [TK]D-Fender | shamess, then you'd need something much closer to home.... |
| 00:19.35 | plik | a number of others do free 0844/5 or 0870, but then the cost per call is higher |
| 00:20.31 | shamess | Yeah. A geographic number would be awesome, since then it'd use my free minutes. But I doubt there are people supplying those for fairly cheap, ne? |
| 00:21.26 | [TK]D-Fender | shamess, Its a question of costs and needs. Who exactly would be using this "service" anyways? |
| 00:21.39 | plik | shamess: callwithus also do uk geo nos foe a few USD per month -good exchange ratestoo see www.callwithus.com otherwise voiptalk.org or voipon.co.uk for a few £ per month |
| 00:21.47 | shamess | Just me. I'm just experimenting. |
| 00:22.40 | equanimity | sipgate do UK geographic numbers for free |
| 00:22.49 | JayTee52 | I "experimented" in college :-) |
| 00:23.06 | plik | oh cool equanimity , thanks |
| 00:23.08 | ManxPower | JayTee52: liked girls more? |
| 00:23.24 | JayTee52 | way, way, way more |
| 00:24.02 | JayTee52 | but whatever floats anyone else's boat is cool with me, I'm straight but not narrow |
| 00:24.18 | asdx | is the [foo] definition also the same as 'username' ? |
| 00:24.21 | asdx | in my iax.conf |
| 00:24.43 | equanimity | plik, no probs :-) |
| 00:25.56 | JayTee52 | god, I got Rickrolled by the Muppets! will this madness never end? |
| 00:26.31 | [TK]D-Fender | shamess, then start with jsut the softphone method and see what expands |
| 00:26.45 | shamess | Btw, for * for what I want, will my server need any hardware it's not likely to have? |
| 00:27.02 | [TK]D-Fender | shamess, For these methods, no. |
| 00:27.03 | UnixDog | ok card listed on ebay |
| 00:27.03 | shamess | Yeah, I will, thanks, [TK]D-Fender. :) |
| 00:27.07 | ManxPower | asdx: sort of. At least you can think of it that way |
| 00:27.14 | ManxPower | well, for incoming calls at least |
| 00:27.16 | JayTee52 | UnixDog, what card? |
| 00:27.56 | UnixDog | sangoma a200d |
| 00:28.09 | *** join/#asterisk rcy (n=rcy@209.90.176.15) |
| 00:28.11 | UnixDog | 2 fxs 2fxo + hwd echo cancler |
| 00:30.16 | JayTee52 | 300 and 500 Buy it now? |
| 00:30.19 | ManxPower | I should start selling prewired Tellabs cards w/echo can cards and power supplies. |
| 00:30.26 | UnixDog | yes |
| 00:31.04 | UnixDog | it would not let me put or best offer |
| 00:31.07 | UnixDog | grrr |
| 00:31.48 | JayTee52 | UnixDog, can't you set a reserve and make the open bid lower than that? Kinda works the same way. |
| 00:32.17 | UnixDog | I tried it would not let me I can try to revise it let me see |
| 00:32.24 | *** join/#asterisk m4sk4r4 (n=m4sk4r4@189.13.182.58) |
| 00:32.43 | JayTee52 | it may not be an option if you use the Buy It Now option. |
| 00:34.19 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 00:34.54 | UnixDog | with buy it now you cant change |
| 00:35.04 | JayTee52 | figures |
| 00:35.29 | JayTee52 | you know you've had a bad day when the elephants have crapped right in front of the steps to your office. |
| 00:36.07 | plik | is there a limit to how long an extension cord can be from and analogue phone to an ATA (prolly a PAP2 or SPA2000, if it matters)? |
| 00:36.16 | JayTee52 | most of the keepers like me but I think they don't like my boss or the rest of the I.T. dept and make them crap there. |
| 00:36.43 | UnixDog | lol |
| 00:36.44 | JayTee52 | plik, 25 to 50' shouldn't be a problem |
| 00:37.10 | plik | sweet, that should do :) |
| 00:37.46 | JayTee52 | I actually have analog lines that are about half a mile from the PBX punchdowns to the phone. |
| 00:37.56 | UnixDog | well I hope the card sells |
| 00:38.08 | JayTee52 | it's not a bad price |
| 00:38.22 | *** join/#asterisk jfg_ (n=jfg@dyn-91-165-217-100.ppp.tiscali.fr) |
| 00:38.46 | JayTee52 | I need to list my Ovation guitar on Ebay. |
| 00:41.55 | [hC] | [TK]D-Fender: hey, what option to use for impossible dial plan matching in your polycom configs? |
| 00:49.09 | *** join/#asterisk ectospasm (n=ectospas@c-71-207-229-248.hsd1.al.comcast.net) |
| 00:49.30 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-e76de24b6107fadc) |
| 00:50.42 | shamess | *cough* So, I just did apt-get install asterisk, and I can't find any information on where to go from there... (Sorry for being tedious.) |
| 00:52.00 | sysreq | ~whatnow |
| 00:52.01 | jbot | i guess now is a good time to tell you that I have 6 gigabytes of data |
| 00:53.08 | sysreq | oh, wrong one. |
| 00:53.12 | sysreq | ~nowwhat |
| 00:53.12 | jbot | So you just installed Asterisk now what? http://www.youtube.com/watch?v=ULgwbvj768E |
| 00:53.14 | JayTee52 | shamess, did you download the pdf version of THE BOOK? |
| 00:53.18 | sysreq | there we go. |
| 00:54.16 | JayTee52 | shamess, also checkout www.asteriskguru.com, www.voip-info.org. Lotta howto's and tutorials there. |
| 00:54.59 | JayTee52 | ~book |
| 00:55.00 | 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, or see ~buybook |
| 00:55.14 | JayTee52 | and that's THE BIBLE. download the free pdf version. |
| 00:57.27 | *** part/#asterisk ThatKidKel (n=Kelvin@lacasa.thatkidkel.net) |
| 00:58.50 | *** join/#asterisk DrkShadow (n=chatzill@host-72-175-240-62.static.bresnan.net) |
| 00:58.56 | DrkShadow | Does anyone have experience with DND on an Aastra phone? There's a soft button for it, but it doesn't cause an Asterisk event like setting DND in the dialplan would do.. |
| 01:10.34 | lirakis | DrkShadow: dnd on the phone is not related to the feature code for dnd in * ... |
| 01:10.37 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 01:11.10 | lirakis | DrkShadow: that function is implemented IN the phone... ususally people have to set "disable call features" in the phone if they want to do stuff completely through asterisk |
| 01:11.47 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 01:13.10 | DrkShadow | hmm. |
| 01:13.31 | JayTee52 | dnd? |
| 01:13.38 | JayTee52 | dialed number display? |
| 01:13.42 | DrkShadow | that's what I feared. I'd like a way for someone to be able to tell if someone's on DND, without having to call them and get a busy signal.. |
| 01:13.50 | lirakis | JayTee52: do not disturb |
| 01:14.00 | JayTee52 | oh, duh! *headdesks* |
| 01:14.11 | lirakis | DrkShadow: disable call features in the phone.. and make people use the feature codes |
| 01:14.18 | lirakis | DrkShadow: then you will know ;) |
| 01:14.33 | DrkShadow | lirakis: then they need to do it for each line, and there's no phone indicator that they're on dnd :-/ |
| 01:14.41 | DrkShadow | can't trust the users.. ;-) |
| 01:14.57 | lirakis | DrkShadow: you dont have to |
| 01:15.03 | lirakis | DrkShadow: --> DISABLE |
| 01:15.08 | lirakis | DrkShadow: call features in the phone |
| 01:15.12 | DrkShadow | disable? |
| 01:15.16 | lirakis | so that hitting the dnd button does jack |
| 01:15.17 | JayTee52 | nope, can't trust the users....they are dirty, evil little things. |
| 01:15.28 | lirakis | DrkShadow: disable .. you know.. to make not work good |
| 01:15.32 | DrkShadow | right. disable dnd. But then they can't see that they are on dnd. And hitting the button only does it on one line.. |
| 01:15.41 | lirakis | DrkShadow: omg |
| 01:16.04 | *** join/#asterisk colinm_ (n=colinm@VDSL-130-13-122-158.PHNX.QWEST.NET) |
| 01:16.05 | lirakis | patience wanes as the day goes on |
| 01:16.16 | DrkShadow | lirakis: I'm not going to track each sip id's to see which are associated with what phone, and set a marker for each sip id when someone types in a * code on one of the lines. |
| 01:16.32 | JayTee52 | I had patience once, but then she moved away to another town. So did hope |
| 01:17.30 | lirakis | DrkShadow: .. use *'s call features and some thing like the flash panel or some thing |
| 01:17.37 | lirakis | DrkShadow: it will show you what everyone is doing |
| 01:17.55 | lirakis | DrkShadow: http://www.asternic.org/ |
| 01:17.57 | DrkShadow | in which case they have to dial a *78 on each line associated with their phone... |
| 01:18.58 | lirakis | DrkShadow: .. yeah .. so? |
| 01:19.11 | DrkShadow | lirakis: ... right.. |
| 01:19.14 | lirakis | DrkShadow: do most of your users have 10 lines? |
| 01:20.14 | DrkShadow | many of them have one, a few have three, some don't know which lines are on asterisk and which are still with Vocalocity, and if they put it on DND using *78, they can't see that it's on DND.. and so I'm willing to bet will not remember to take it off as they pass their desk and pick up their headset when they get back from wherever they were. |
| 01:20.19 | DrkShadow | I don't trust the users. |
| 01:20.53 | lirakis | DrkShadow: depending on what the phone is .. you can set sip notify to change screen settings etc. |
| 01:21.00 | lirakis | DrkShadow: but youve got to pick your battles |
| 01:21.17 | DrkShadow | well.. not on this aastra. I'm just gonna say: you won't know if someone's set to DND. |
| 01:22.28 | lirakis | DrkShadow: dude... YOU would know .. from some AMI interface like FOP, and if it has a graphic display of some kind you can tell * to change notifications on the screen's display |
| 01:23.20 | DrkShadow | if I use the phone's DND, there is no notification on the server. If I use an asterisk code, there is no notification on the phone. I'm not going to police the entire company when they get back from lunch. |
| 01:23.25 | DrkShadow | uhm.. |
| 01:23.34 | DrkShadow | dunno how to change the phone's display in such a way. |
| 01:23.43 | DrkShadow | nor if I can on other models (grandstreams) |
| 01:23.49 | lirakis | DrkShadow: ARG!! you arent freakin listening |
| 01:23.52 | DrkShadow | but.. interesting. Something to look into. |
| 01:23.55 | lirakis | walks away |
| 01:24.00 | DrkShadow | lirakis: I reread and corrected myself ;-) |
| 01:27.25 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
| 01:30.09 | ManxPower | DrkShadow: 1.6 has support for changing the BLF state of LEDs |
| 01:30.21 | ManxPower | devstate or something like that |
| 01:30.28 | DrkShadow | actually.. since this system is in and working, I will probably never upgrade it. |
| 01:30.37 | DrkShadow | but.. interesting ;-) |
| 01:34.14 | *** join/#asterisk BeeBuu (n=beebuu@218.13.83.47) |
| 01:35.26 | JayTee52 | I'm very interested in 1.6 because it has SIP/TCP |
| 01:36.25 | JayTee52 | and that'll mean * can talk directly to my Exchange Unified Messaging instead of going through a sipX router. |
| 01:39.52 | DrkShadow | what is sipx? I thought it was sipXecs, an asterisk distro.. |
| 01:40.22 | JayTee52 | no, it's not * |
| 01:40.44 | DrkShadow | is it sipXecs? |
| 01:40.48 | ManxPower | JayTee52: I plan on skipping 1.4 all togather . |
| 01:41.26 | JayTee52 | DrkShadow, sipXecs is the full enterprise version that you buy if I'm not mistaken. |
| 01:42.58 | JayTee52 | it can be it's own VOIP PBX but doesn't support zaptel so you can't put cards in the server, you'd have to use appliances that convert can do FXO to TDMoE etc. |
| 01:43.58 | JT | iirc they renamed from sipX to sipXecs |
| 01:44.02 | JayTee52 | we just use the open source version of sipX for our testing for Exchange Unified Messaging because it will do SIP UDP to SIP TCP transforms. |
| 01:44.32 | JayTee52 | JT, you may be right. It's been several months since I've been to their site. |
| 01:44.46 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 01:44.46 | *** mode/#asterisk [+o lmadsen] by ChanServ |
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| 01:48.13 | keith4_ | so... I kinda like Snap (snapanumber), but I really dislike Windows. Is there any hope? |
| 01:49.00 | JayTee52 | with Windows it's usually "Live in hope, die in despair" |
| 01:52.20 | keith4_ | I found this thing, for OS X: http://click.rho.cc/ |
| 01:52.27 | keith4_ | how about a linux solution? |
| 02:10.24 | *** join/#asterisk nhuismanwork (n=nhuisman@dhcp18.IfA.Hawaii.Edu) |
| 02:10.36 | nhuismanwork | is it possible to setup a cisco 7940 with static ip settings? |
| 02:10.40 | nhuismanwork | and not need to use a dhcp/tftp server? |
| 02:11.06 | nhuismanwork | i'm trying to get a phone to register from a remote network |
| 02:11.41 | nhuismanwork | it won't let me statically set the ip/network settings on the phone |
| 02:11.56 | nhuismanwork | is the solution to set them up via tftp at the local site and then at the remote site it will just keep the settings? |
| 02:12.39 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 02:19.32 | *** join/#asterisk Rias (n=VANDORGE@216-250-186-217.static.iphouse.net) |
| 02:22.10 | [hC] | so polycom people, have any of you noticed that the way the cursor position on the screen looks now in firmware 3.0 looks different? |
| 02:24.18 | [TK]D-Fender | [hC], how so? |
| 02:25.29 | [hC] | When i dial, I get a reverse-contrast box around the number ive just dialed, until i wait 2 seconds, then the cursor goes back to being in front of the latest number |
| 02:25.36 | [hC] | it looks like insert-mode in a text editor |
| 02:25.46 | [hC] | And I dont think it used to do that. |
| 02:27.15 | [hC] | Im wondering.. I think the way I am doing overrides is screwing up some of the config i want the phone to actually keep. |
| 02:27.34 | [hC] | For example, in sip.cfg, there's a line, <call var1=value var2=value var3=value var4=value etc> |
| 02:27.49 | [hC] | I include my own config overrides, and in my file i just have <call oneoption=itsvalue> |
| 02:27.58 | [hC] | Does this mean var1, var2, var3, are ignored essentially? |
| 02:33.39 | *** join/#asterisk Synoptic (i=Synoptic@modemcable105.136-203-24.mc.videotron.ca) |
| 02:33.41 | Synoptic | hi there |
| 02:34.02 | Synoptic | Why would my tdm400p have low sounf on receive ? (i.e I cannot hear loud, but people hear me good) |
| 02:34.15 | [hC] | rxgain is too low in zapata.conf |
| 02:34.19 | [hC] | increase it by 3 |
| 02:34.22 | Synoptic | ok |
| 02:34.33 | Synoptic | and how do I reset to test after ? |
| 02:34.41 | [hC] | restart asterisk and try another call |
| 02:34.55 | Synoptic | ok |
| 02:34.56 | Synoptic | well |
| 02:35.03 | Synoptic | I've tried already from 0 to 8 |
| 02:35.06 | Synoptic | no change |
| 02:35.39 | Synoptic | maybe I have somethign else in my config that renders the settings unused ? |
| 02:35.50 | [hC] | did you fully restart asterisk or just reload? |
| 02:36.10 | [hC] | you can either fully restart asterisk, or doing a reload chan_zap.so will also force the gains to be updated |
| 02:36.15 | [hC] | but simply doing 'reload' will not adjust the gains. |
| 02:36.28 | Synoptic | well |
| 02:36.33 | Synoptic | I did shutdown asterisk |
| 02:36.35 | Synoptic | and restart it |
| 02:36.40 | Synoptic | let me try the reload module.. |
| 02:36.52 | [hC] | fully restarting asterisk should do it as well |
| 02:37.08 | [hC] | so you already went from rxgain=0.0 to rxgain=8.0 and there was no effect on the volume you heard? |
| 02:38.09 | asdx | how do i eliminate echo/noise |
| 02:38.19 | [hC] | on an analog line? |
| 02:38.24 | asdx | digital |
| 02:38.26 | asdx | internet |
| 02:38.38 | asdx | asterisk/iax |
| 02:38.41 | Synoptic | [hC] : exactly.. now I went for 8 to 20 |
| 02:38.44 | [hC] | well echo can only be introduced in an analog portion of a call |
| 02:38.45 | Synoptic | no difference |
| 02:39.09 | [hC] | Synoptic: you must be missing something, that would make a very big difference. |
| 02:39.17 | [hC] | Synoptic: pastebin your zapata.conf at pastebin.ca |
| 02:40.04 | Synoptic | http://www.pastebin.ca/976353 |
| 02:40.06 | Synoptic | there you go |
| 02:41.02 | [hC] | you're going to want to move channel => 1 to the end of the file |
| 02:41.08 | Synoptic | ok |
| 02:41.13 | [hC] | actually |
| 02:41.14 | Synoptic | order is imprtant ? |
| 02:41.17 | [hC] | yes |
| 02:41.19 | Synoptic | ahhh |
| 02:41.22 | Synoptic | me should read more |
| 02:41.26 | [hC] | move group=0 to the second to last line, and channel=1 to the last line |
| 02:41.32 | [hC] | otherwise the options below it wont do anything |
| 02:41.52 | Synoptic | group=0 just before channel=1 ? |
| 02:41.57 | [hC] | yeah |
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| 02:42.17 | *** join/#asterisk _mwoodj_ (n=MWoodJ@pdpc/sponsor/digium/hyper-eye) |
| 02:42.38 | Synoptic | trying |
| 02:43.22 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 02:44.34 | *** join/#asterisk gerphimum (n=trekkie@cpe-70-125-151-117.satx.res.rr.com) |
| 02:45.13 | Synoptic | ok |
| 02:45.22 | Synoptic | txgain seems to work :) |
| 02:45.24 | Synoptic | rx soso |
| 02:45.29 | Synoptic | but it now makes some difference |
| 02:45.58 | Synoptic | what is the maximium allowd gain ? |
| 02:45.58 | [hC] | yeah you might have to adjust it bit by bit to get it to where you need it but you should notice a difference now |
| 02:46.02 | [hC] | 100 i think |
| 02:46.06 | Synoptic | k |
| 02:46.09 | [hC] | but personally ive never gone over like.. 25 |
| 02:46.11 | Synoptic | thnks a lot |
| 02:46.18 | [hC] | no problem |
| 02:46.28 | Synoptic | i'll try to get some distrotion just to see if I get any difference |
| 02:47.20 | Synoptic | ahahahahaha |
| 02:47.22 | Synoptic | ok |
| 02:47.28 | Synoptic | setting it to 80 = SCREAMS |
| 02:47.41 | Synoptic | it screams like the death ahaha |
| 02:54.13 | [hC] | now how the heck does a polycom know to allow dialing *codes if there is no *anything in the dial plan digitmap. |
| 02:54.56 | [TK]D-Fender | [hC], it doesn't, which is why you ADD IT |
| 02:55.51 | [hC] | [TK]D-Fender: I do not have it, and it is letting me dial *codes, however after pressing * once, it returns back to a dial tone and asks for more digits. Hence why i asked! |
| 02:55.56 | [hC] | but, it does WORK |
| 02:56.33 | drmessano | Because a * is a single digit and being trapped by a rule for one digit? |
| 02:57.25 | [TK]D-Fender | [hC], because of your impossiblematch handling |
| 02:57.36 | [hC] | [TK]D-Fender: ah, you're right. |
| 02:57.37 | [TK]D-Fender | [hC], I advise "2" |
| 02:57.48 | drmessano | or by dialing 1234567890 it expects something else.. and * and # it doesnt? |
| 02:57.50 | drmessano | Oh |
| 02:57.54 | [TK]D-Fender | [hC], aka the "STFU and just do what I say" match |
| 02:58.12 | [hC] | I am currently using 0, I was trying to test if it matched each digit against the live dial plan on the sip server, rather than forcing the use of the one on the phone. |
| 02:59.36 | [hC] | Ive added * to my dial plan, and will try out impossible match handling set to 2. |
| 03:00.18 | [hC] | Id love to know if my includes are clobbering defaults by leaving them out or not, but its kinda hard to tell. |
| 03:05.06 | drmessano | Liek, omg |
| 03:16.00 | *** join/#asterisk tengulre11 (n=tengulre@124.42.50.9) |
| 03:16.59 | *** join/#asterisk micander (n=Michael_@ip70-181-134-119.sd.sd.cox.net) |
| 03:27.19 | *** join/#asterisk blq (n=Bl@dslb-088-066-237-234.pools.arcor-ip.net) |
| 03:35.13 | *** part/#asterisk fluxp (i=flux@hanso.tv) |
| 03:35.34 | *** join/#asterisk bintut (n=bintut@bb116-15-101-181.singnet.com.sg) |
| 03:37.28 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
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| 04:19.48 | *** join/#asterisk fiftysixer (n=fiftysix@adsl-76-233-74-185.dsl.stlsmo.sbcglobal.net) |
| 04:19.53 | *** part/#asterisk fiftysixer (n=fiftysix@adsl-76-233-74-185.dsl.stlsmo.sbcglobal.net) |
| 04:19.54 | *** join/#asterisk fiftysixer (n=fiftysix@adsl-76-233-74-185.dsl.stlsmo.sbcglobal.net) |
| 04:20.11 | *** join/#asterisk lowlevel (n=Stuart@lowlevel.ca) |
| 04:21.10 | *** join/#asterisk mmurdock (n=blah@c-98-202-154-226.hsd1.ut.comcast.net) |
| 04:22.11 | mmurdock | Is there a way to have a use hear a different ring when the person they are calling is already taking a call? |
| 04:23.03 | [TK]D-Fender | murdock : with excessive hacing, yes |
| 04:23.34 | mmurdock | not with dialplan magic then. |
| 04:23.57 | [TK]D-Fender | mmurdock, use Chanisavail to see if they're on a call if so change the inidication zone, Answer the call and dial them with "r" to force the indication as audio. |
| 04:24.05 | [TK]D-Fender | mmurdock, Actually, scratch that, easier still |
| 04:24.27 | mmurdock | I know how to change the callee's ringer. |
| 04:24.31 | [TK]D-Fender | mmurdock, If Chanisavail say they're busy, use "M" to play MoH of a class with the audio of the tone you want them to hear |
| 04:25.07 | mmurdock | Or if then number of channels in use is greater then 1. |
| 04:25.18 | mmurdock | ? |
| 04:25.27 | [TK]D-Fender | mmurdock, Same thing, but the MoH approach might be the cleanest |
| 04:26.07 | *** join/#asterisk jameswf-home (n=james@ip72-204-221-181.ph.ph.cox.net) |
| 04:26.46 | mmurdock | Does Chanisavail say busy if there call limit is greater then 1 and they are already taking a call? |
| 04:27.40 | jameswf-home | I think chanunavail is reorder |
| 04:32.02 | *** join/#asterisk obnauticus (n=obnautic@c-67-160-183-109.hsd1.wa.comcast.net) |
| 04:33.09 | [TK]D-Fender | mmurdock, use the "s" option. |
| 04:36.29 | mmurdock | s option? |
| 04:37.11 | mmurdock | extension? |
| 04:38.56 | *** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr) |
| 04:39.46 | [TK]D-Fender | mmurdock, "core show application chanisavail" |
| 04:40.17 | *** join/#asterisk JT (n=j@unaffiliated/jt) |
| 04:40.21 | jameswf-home | heh http://xkcd.com/400/ |
| 04:41.03 | mmurdock | [TK]D-Fender>: Sorry, I wasn't sure what app you were refering to. |
| 04:41.56 | [TK]D-Fender | mmurdock, The only one I mentioned. Would seem pretty obvious... |
| 04:43.17 | mmurdock | [TK]D-Fender>: For some reason I was looking a the moh stuff. |
| 04:43.19 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:44.05 | *** join/#asterisk mandd (n=dache@dsl-135-236.aei.ca) |
| 04:44.12 | mandd | there, freebsd 7.0 |
| 04:44.28 | mandd | no more isa |
| 04:52.19 | jameswf-home | mandd: recompile |
| 04:53.10 | mandd | recompile? |
| 04:53.18 | mandd | recompile what? |
| 04:53.30 | mandd | was compiling for hours now |
| 04:53.33 | mandd | no more |
| 04:53.41 | mandd | last buildworld and thats it |
| 05:00.07 | *** join/#asterisk yxa (n=lonari@58.185.90.101) |
| 05:08.17 | jameswf-home | ~bsd |
| 05:08.18 | jbot | BSD is a UNIX operating system. An asterisk port is currently availible if you feel you must, or a way to set your pc back 30 years, progress is overrated |
| 05:09.14 | *** join/#asterisk steliosk (n=Stelios@85.72.79.254) |
| 05:13.33 | mandd | jameswf-home nah, running * on ubuntu, bsd is routing |
| 05:13.54 | mandd | had windows server/w ISA, and it messed with sip packets |
| 05:14.49 | *** join/#asterisk Nasra (n=Nasra@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 05:21.16 | jblack | bsd is still out there? |
| 05:30.03 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-82a603ffac2ac474) |
| 05:33.35 | *** join/#asterisk sbs_a (n=Sarin@117.196.133.173) |
| 05:36.19 | *** join/#asterisk Vco (n=Vco@S010600095bdbc09f.cg.shawcable.net) |
| 05:41.39 | styelz | i installed * on a openbsd box a bit ago |
| 05:42.04 | styelz | good for sip. no zap support |
| 05:43.02 | asdx | is iax a standard already |
| 05:43.09 | BBHoss | heh |
| 05:43.58 | JT | depends what you define as a standard |
| 05:45.00 | asdx | is there many devices out there that already support the protocol? |
| 05:48.01 | JT | not really |
| 05:48.07 | JT | i wouldn't expect that to change |
| 05:48.21 | asdx | why not |
| 05:48.42 | asdx | if is a better protocol, why not support it |
| 05:48.47 | *** join/#asterisk sbs_a (n=Sarin@117.196.133.173) |
| 05:49.08 | JT | it's not a better protocol |
| 05:49.14 | JT | and pretty much only asterisk supports it |
| 05:49.31 | JT | it makes no sense for a hardware vendor to produce a device which is less supported |
| 05:49.32 | asdx | well, i like it better than SIP, at least i don't have to deal with NAT issues |
| 05:49.38 | JT | and therefore giving them less sales |
| 05:49.48 | JT | i rarely have NAT issues with SIP |
| 05:50.58 | JT | also, there's almost no service provider support for IAX |
| 05:51.02 | JT | because it does not scale |
| 05:55.11 | asdx | hm |
| 05:56.56 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 06:13.08 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 06:13.08 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 06:14.50 | *** join/#asterisk Kernel_Core (n=I@213.207.243.227) |
| 06:16.52 | Kernel_Core | hi all |
| 06:18.27 | *** join/#asterisk drehlecom (i=ircbnc@2001:6f8:1153:2:208:c7ff:feac:d1fb) |
| 06:18.28 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:21.31 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
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| 06:23.44 | *** join/#asterisk Alpha_AI (n=Ben@210.11.97.57) |
| 06:26.40 | Alpha_AI | Hello, I know someone who is selling an IBM xSeries 366 for $400. It comes with 4x XEON MP 3.6ghz CPU and 16GB memory |
| 06:26.45 | Alpha_AI | is that a great price? |
| 06:28.24 | JT | seems alright, considering the ram |
| 06:29.06 | *** join/#asterisk drcode (n=user1@85.65.12.156.dynamic.barak-online.net) |
| 06:29.19 | drcode | hi all |
| 06:29.20 | drcode | whats up |
| 06:29.29 | drcode | I need help with SIP and QOS |
| 06:29.41 | BBHoss | drcode, sup dog |
| 06:29.45 | drcode | I have SIP client behind my linux router /server |
| 06:29.51 | drcode | I want to give this SIP QOS |
| 06:30.05 | drcode | (its FXO device) |
| 06:30.22 | drcode | any idea how I can do QOS under my linux server? |
| 06:30.40 | *** part/#asterisk kumamoto (n=eryco@66-169-124-120.dhcp.ftwo.tx.charter.com) |
| 06:30.43 | BBHoss | by fxo device you mean an ata? |
| 06:32.30 | BBHoss | drcode, what kind of QoS do you want to do? |
| 06:33.26 | *** join/#asterisk bintut (n=bintut@bb116-15-101-181.singnet.com.sg) |
| 06:40.47 | *** join/#asterisk infamy (n=infamy@216.19.180.122.novuscom.net) |
| 06:41.16 | infamy | Where can i normally find the startup log for asterisk... (trying to figure out why i cannot get the static-html webserver to work |
| 06:44.25 | *** join/#asterisk Kumbang (n=dsp@167.205.24.69) |
| 06:45.29 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 06:49.15 | *** join/#asterisk Al_WinKiller (n=winkille@mgm.cornet.am) |
| 06:49.57 | Al_WinKiller | hi guys I have setup conference room but in debug I get this message "pbx.c:1821 pbx_extension_helper: No application 'MeetMe' for extension (cornet, 900, 1)" |
| 06:50.04 | Al_WinKiller | any suggestion ? |
| 06:52.07 | BeeBuu | how can i dial out in AGI? |
| 07:00.26 | Aces1up | anyone here used DIdexchange? i am trying to find a wholesale DID provider.. that will allow me to use DID's for calling cards how do i find these number in didexchange |
| 07:00.36 | *** join/#asterisk shinao1 (n=shinao1@41.221.165.43) |
| 07:00.58 | *** join/#asterisk mikefut (n=mikefut@lan.integra-media.com) |
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| 07:12.13 | *** join/#asterisk bougie (n=bougie@APoitiers-256-1-31-46.w90-16.abo.wanadoo.fr) |
| 07:12.28 | bougie | hello :) |
| 07:17.29 | *** join/#asterisk solar_ant (n=John@61.17.177.5) |
| 07:17.51 | *** join/#asterisk xenonex (n=xenonex@82.200.211.1) |
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| 07:33.30 | *** join/#asterisk Sir0x (n=cmayer@p5B0842DB.dip.t-dialin.net) |
| 07:33.36 | Sir0x | #join asterisk-dev |
| 07:35.51 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 07:36.00 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 07:36.04 | marc7 | that's a great channel |
| 07:37.45 | *** join/#asterisk xenonex (n=xenonex@82.200.211.1) |
| 07:42.28 | *** join/#asterisk oej (n=olle@bkkb-gw.voop.net) |
| 07:46.25 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 07:47.23 | *** join/#asterisk mikkel (n=mikkel@130.226.36.170) |
| 08:00.48 | *** join/#asterisk qdk (n=qdk@85.235.253.139) |
| 08:11.08 | *** join/#asterisk NirS (n=Nir@212.150.121.71) |
| 08:11.11 | NirS | hi all |
| 08:11.19 | NirS | how are we doing this fine morning ? |
| 08:12.43 | *** join/#asterisk Tili (n=tili@58.27.147.108) |
| 08:14.22 | Sir0x | hi Nirs, fine thank just a little bit tired. |
| 08:14.33 | NirS | tired is good |
| 08:14.38 | NirS | it means you are keeping busy |
| 08:15.22 | *** join/#asterisk jivco (n=jivco@85.187.217.6) |
| 08:16.09 | Sir0x | LOL, that's a matter of opinion. |
| 08:16.10 | NirS | say, anyone using RNK as their termination partner for Asterisk ? |
| 08:17.00 | NirS | I'm trying to setup Asterisk with RNK, and I'm running into issues |
| 08:17.06 | NirS | I wonder what can be wrong |
| 08:18.36 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 08:18.54 | Al_WinKiller | what is up ? |
| 08:21.52 | *** join/#asterisk stimpie (n=stimpie@84-104-6-10.cable.quicknet.nl) |
| 08:23.25 | NirS | say, anyone ever utilized Asterisk SIP when SIP is performed to one IP, while the RTP goes to another IP ? |
| 08:23.31 | NirS | should I use the outbound proxy setting ? |
| 08:25.27 | *** join/#asterisk rulereric (n=esnellma@ip70-185-121-45.ga.at.cox.net) |
| 08:32.15 | *** join/#asterisk hohum (n=dcorbe@99-203-98-1.area1.spcsdns.net) |
| 08:33.18 | *** join/#asterisk Dr-Linux (n=somethin@117.20.21.66) |
| 08:34.16 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 08:35.59 | Pagautas | hi |
| 08:36.18 | Pagautas | is it possible to execute sql code in dialplan? |
| 08:36.43 | Pagautas | like select 1 from allow where number='${CALLERID(number)}' |
| 08:40.22 | *** join/#asterisk xenonex (n=xenonex@89.218.237.27) |
| 08:40.26 | Sir0x | hi pagautas, which database do you use ? |
| 08:40.41 | *** part/#asterisk Strom_M (n=strom@208.127.172.112) |
| 08:43.12 | Pagautas | mysql |
| 08:43.33 | Sir0x | there is a application you can use in your dialplan named MYSQL |
| 08:43.40 | Sir0x | http://www.voip-info.org/wiki/view/Asterisk+cmd+MYSQL |
| 08:43.57 | *** join/#asterisk radulovity (n=emil@77.46.189.139) |
| 08:44.02 | radulovity | Morning ... |
| 08:44.06 | Sir0x | morning |
| 08:44.40 | radulovity | I have configured askozia asterisk based VoIP distro, and if i try to dial over my IAX2 softphone program, become i next error messsage |
| 08:44.47 | radulovity | http://www.pastebin.ca/976576 |
| 08:45.26 | radulovity | But, i known that is this line free, and that on another side is the line free (free information number about time and road states) |
| 08:45.31 | radulovity | Why? :) |
| 08:45.36 | *** join/#asterisk oej_ (n=olle@bkkb-gw.voop.net) |
| 08:46.32 | Pagautas | Sir0x: thanks, its what i was looking for |
| 08:46.56 | Sir0x | you're welcome |
| 08:49.43 | *** join/#asterisk denzs (n=denzs@carbon.gonicus.de) |
| 08:51.11 | denzs | hi, can you tell if it's possible to use realtime switches inside of macros in ael? i already tried the "eswitches" statement which works great in a normal context, but it doesnt work in a macro ... |
| 08:55.57 | NirS | oej_, you there ? |
| 08:57.04 | NirS | anyone using RNK as a SIP termination provider ? |
| 09:04.54 | *** join/#asterisk tinapa (n=niwre@67.159.178.21) |
| 09:06.45 | tinapa | hello, can anyone give ideas on how to create an auto dialer with asterisk? |
| 09:07.03 | tinapa | or any tutorial that can help |
| 09:08.26 | *** join/#asterisk lesouvage (n=lesouvag@217.166.84.137) |
| 09:11.02 | *** join/#asterisk ptblank (n=MURDER1@cpe-75-84-47-107.socal.res.rr.com) |
| 09:11.46 | *** join/#asterisk Chris-NB (n=chris@213162066149.public.t-mobile.at) |
| 09:13.28 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 09:14.49 | NirS | tinapa, what do you mean by auto dialer? are you referring to a power dialer? or simply a click-2-call interface ? |
| 09:15.51 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 09:16.25 | tinapa | NirS, a predictive dialer for a contact center |
| 09:22.43 | Dr-Linux | any macro expert active? :) |
| 09:23.26 | *** join/#asterisk nixguy (n=matmoj@fw.packetfront.com) |
| 09:26.39 | nixguy | hi i want to set hide callerid |
| 09:27.00 | nixguy | but alas i fail |
| 09:27.06 | nixguy | usecalledid=yes |
| 09:27.11 | NirS | tinapa, writing a predictive dialer is a work of art |
| 09:27.20 | nixguy | hidecallerid=yes |
| 09:27.21 | NirS | not something that can be done in a few days time |
| 09:27.27 | nixguy | but it still shows up! |
| 09:27.30 | NirS | how many agents you have in your call center ? |
| 09:29.57 | Dr-Linux | what does this line means: |
| 09:29.58 | Dr-Linux | exten => s-NOANSWER,n,GotoIf($["${DTIME}" = "0"]?4) |
| 09:30.23 | Dr-Linux | upon the NO Answer GoTO ?? |
| 09:30.30 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 09:39.38 | JT | NirS: Set(CALLERID(num)=) |
| 09:39.39 | *** join/#asterisk jarod14 (n=jarod14@LMontsouris-152-63-1-19.w80-12.abo.wanadoo.fr) |
| 09:39.41 | JT | err |
| 09:39.44 | JT | nixguy: |
| 09:39.49 | JT | nixguy: depends on channel type too |
| 09:40.53 | jarod14 | Hi all, is there someone, speaking dutch. I need to cut dutch prompts but I do not understand dutch. |
| 09:44.08 | *** join/#asterisk solar_ant (n=John@61.17.177.5) |
| 09:44.21 | *** join/#asterisk hedenberg (n=hedenber@user238.85-195-7.netatonce.net) |
| 09:46.40 | hedenberg | Hi. Is it possible to record two files simuntaneously on the same extension in asterisk? (one for the full conversation, and one started and stopped at demand) |
| 09:48.11 | *** part/#asterisk Sir0x (n=cmayer@p5B0842DB.dip.t-dialin.net) |
| 09:55.58 | *** join/#asterisk steliosk (n=Stelios@athedsl-282332.home.otenet.gr) |
| 09:56.20 | *** join/#asterisk af_ (n=getsmart@88-149-240-26.dynamic.ngi.it) |
| 09:56.51 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.140) |
| 09:58.24 | *** join/#asterisk iunixan (n=iunixan@41.234.141.122) |
| 10:01.49 | iunixan | is it possible to use Panasonic kx-ta308 PBX with Asterisk via it's serial port? |
| 10:03.21 | hedenberg | Hi. Is it possible to record two files simuntaneously on the same extension in asterisk? (one for the full conversation, and one started and stopped at demand) |
| 10:04.19 | BeeBuu | why the log file show "ANSWERED" when i do not pick up? |
| 10:04.35 | BeeBuu | "","","888","call-out","","Zap/3-1","Zap/2-1","Dial","zap/1/802&zap/2/802||Tr","2008-04-08 09:55:09","2008-04-08 09:55:15","2008-04-08 09:55:20",11,5,"ANSWERED","DOCUMENTATION","1207648509.4","" |
| 10:04.44 | BeeBuu | anyone help me please? |
| 10:04.45 | JT | BeeBuu: POTS? |
| 10:05.07 | BeeBuu | what's going on? |
| 10:05.19 | JT | BeeBuu: Is it POTS? answer the question. |
| 10:05.21 | BeeBuu | asterisk 1.14.10 |
| 10:05.23 | JT | BeeBuu: Is it POTS? answer the question. |
| 10:05.59 | BeeBuu | that's the record in message file |
| 10:06.10 | BeeBuu | what's POTS? |
| 10:06.11 | JT | i know that |
| 10:06.14 | JT | ... |
| 10:06.15 | JT | google it |
| 10:06.19 | JT | ~pots |
| 10:06.19 | jbot | extra, extra, read all about it, pots is Plain Old Telephone Service as in "Old Analogue Crap" |
| 10:06.54 | *** join/#asterisk Sir0x (n=cmayer@p5B0842DB.dip.t-dialin.net) |
| 10:07.03 | BeeBuu | Old Analogue crap? |
| 10:07.16 | JT | BeeBuu: www.google.com |
| 10:07.21 | JT | type in pots |
| 10:07.26 | JT | and find out what it is |
| 10:07.43 | JT | any explanation we give here, you will just answer with another question questioning the answer |
| 10:08.13 | tinapa | plain old telephone system or something? |
| 10:08.18 | JT | service |
| 10:08.30 | tinapa | thats it :) |
| 10:08.33 | tinapa | right |
| 10:08.56 | BeeBuu | Plain old telephone service? |
| 10:09.06 | tinapa | yup |
| 10:09.11 | JT | BeeBuu: right, that's what it is, 10 points! |
| 10:09.17 | tinapa | hahahha |
| 10:09.18 | JT | BeeBuu: are you using it or not? |
| 10:09.22 | BeeBuu | so how to fix my problem? |
| 10:09.36 | JT | BeeBuu: Step 1> Answer my question |
| 10:09.38 | JT | Step 2> |
| 10:09.40 | JT | ?? |
| 10:09.43 | JT | Step 3> Profit! |
| 10:10.26 | BeeBuu | that issue come with Plain old telephone service ? |
| 10:10.33 | JT | yes |
| 10:10.38 | JT | are you using it or not |
| 10:10.43 | JT | this is a boolean question |
| 10:10.45 | JT | yes or no |
| 10:10.47 | BeeBuu | no |
| 10:10.51 | BeeBuu | :-) |
| 10:10.53 | JT | what are you using? |
| 10:11.10 | JT | "Zap/3-1","Zap/2-1" |
| 10:11.18 | JT | what sort of zap technology is it? |
| 10:11.24 | BeeBuu | an old pbx |
| 10:11.31 | JT | err |
| 10:11.34 | JT | that's analogue |
| 10:11.36 | JT | ergo pots |
| 10:11.43 | JT | this problem is a POTS problem |
| 10:11.51 | JT | get digital lines and it will not be a problem |
| 10:12.38 | BeeBuu | analogue does not work? |
| 10:12.50 | JT | analogue doesn't work WELL |
| 10:13.08 | JT | you will have issues such as incorrect accounting |
| 10:13.31 | BeeBuu | i got it.thanks. |
| 10:25.36 | hi365 | Gotoif($[$["${SPEECH_SCORE(0)}" < "800"]["{SPEECH_SCORE(0)}" = ""]]?badscore,1) <-------whats wrong with this line? |
| 10:25.47 | hi365 | the error im getting is: [Apr 8 13:24:13] WARNING[12244] ast_expr2.fl: ast_yyerror(): syntax error: syntax error, unexpected '', expecting $end; Input: |
| 10:25.47 | hi365 | 1["{SPEECH_SCORE(0)}" = ""] |
| 10:28.59 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 10:30.26 | JT | hi365: that just looks wrong |
| 10:30.51 | JT | otoif($[$["${SPEECH_SCORE(0)}" < "800"]["{SPEECH_SCORE(0)}" = |
| 10:30.51 | JT | <PROTECTED> |
| 10:30.56 | JT | oops |
| 10:31.04 | JT | Gotoif($[$["$ |
| 10:32.02 | hi365 | think i got it; $["${SPEECH_SCORE(0)}" < "800"]|$["{SPEECH_SCORE(0)}" = ""] <--- more like it |
| 10:34.34 | Dr-Linux | JT: Hi |
| 10:34.38 | Dr-Linux | exten => s-NOANSWER,n,GotoIf($["${DTIME}" = "0"]?4) |
| 10:34.45 | Dr-Linux | upon the NO Answer GoTO ?? |
| 10:35.45 | nixguy | hidecallerid=yes in zapata.conf doesn't work |
| 10:35.57 | nixguy | anything i might be missing? |
| 10:36.50 | ccesario | hello I see in #asterisk-commits any commits about mfcr2 in 1.4 branch....... it is working ? |
| 10:38.40 | JT | nixguy: didn't i ask you what type of channel it was before? |
| 10:38.54 | nixguy | hm must have missed it sorry |
| 10:38.55 | nixguy | its pri |
| 10:39.00 | nixguy | JT: |
| 10:40.23 | JT | nixguy: try SetCallerPres(prohib) |
| 10:40.36 | nixguy | in my zapata.conf? |
| 10:40.48 | nixguy | or as a part of my trunkconfig? |
| 10:41.05 | nixguy | hmm looks like the syntax for extensions.conf |
| 10:43.37 | nixguy | nope still shows the number of my pbx |
| 10:43.37 | JT | extensions.conf |
| 10:43.45 | nixguy | (added to extensions.conf) |
| 10:43.51 | JT | how are you adding it? |
| 10:44.21 | nixguy | JT: |
| 10:44.23 | nixguy | exten => _0.,1,SetCallerPres(prohib) |
| 10:44.24 | nixguy | exten => _0.,n,Dial,Zap/g1/${EXTEN} |
| 10:44.41 | JT | what version of asterisk? |
| 10:44.44 | nixguy | 1.2 |
| 10:45.13 | JT | sounds like your telco is ignoring the request for blocked callerid |
| 10:45.21 | JT | check that asterisk is actually sending such a request |
| 10:45.27 | JT | with pri intense debug |
| 10:45.32 | nixguy | JT: hmm ok that sounds possible, |
| 10:45.42 | Dr-Linux | JT: can i /msg you? |
| 10:45.50 | JT | Dr-Linux: about? |
| 10:45.57 | Dr-Linux | i wanna discuss something about failover |
| 10:46.23 | JT | is there a reason it can't be done in channel? |
| 10:46.37 | Dr-Linux | JT: sure why not if you are interested |
| 10:47.49 | Dr-Linux | JT: i've two host on SIP[IAX] .. i want if one host is down, the system should automatically send call to the 2nd host |
| 10:48.25 | Dr-Linux | i got something from the web .. but i'm little confused with this .. actually it's being some somthing in macros |
| 10:49.00 | Dr-Linux | JT: should i pastebin it to you, so you will be able to guide? |
| 10:49.06 | Dr-Linux | and discuss further |
| 10:50.29 | Dr-Linux | JT: go here : |
| 10:50.30 | Dr-Linux | http://phpfi.com/308314 |
| 10:50.52 | Dr-Linux | first of all tell me what does line13 means? |
| 10:54.45 | *** part/#asterisk fish_ (n=fish@freigeist.org) |
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| 11:03.47 | nibbler_de | am i doing something terribly wrong by trying to match the pattern "one or more digits" "asterisk" "one or more digits" with _.X*X. in my extensions.conf? |
| 11:08.23 | marc7 | ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected '>=', expecting $end; |
| 11:08.32 | marc7 | what module am I missing? |
| 11:10.26 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 11:21.55 | tzafrir | marc7, what is the line? |
| 11:22.15 | *** join/#asterisk sysreq` (n=sysreq@unaffiliated/sysreq) |
| 11:22.29 | marc7 | exten => _XXXXXX,n(notauth),GotoIf($[${LoginTry} >= 3]?i,denied) |
| 11:22.30 | marc7 | exten => _XXXXXX,n,Set(LoginTry=$[${LoginTry}+1]) |
| 11:24.40 | marc7 | oh it's probably because I'm not setting ${LoginTry} ahead of time |
| 11:24.53 | bougie | anyone knows the address of the french irc channel for asterisk ? |
| 11:24.57 | *** join/#asterisk jfg (n=jfg@dyn-91-165-217-100.ppp.tiscali.fr) |
| 11:27.14 | marc7 | bougie: there's the Asterisk Montreal User Group in #amug |
| 11:27.28 | marc7 | you might find forums or mailing lists a better medium for support in french |
| 11:27.33 | marc7 | bonne chance! |
| 11:28.16 | *** join/#asterisk lesouvage (n=lesouvag@217.166.84.137) |
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| 11:39.43 | Al_WinKiller | hi , anybody around ? |
| 11:42.20 | bougie | thanks marc7 |
| 11:43.22 | *** join/#asterisk jfg (n=jfg@dyn-91-165-217-100.ppp.tiscali.fr) |
| 11:44.20 | Al_WinKiller | guys how can I perform call transfer using "FLASH" button on standard Panasonic phones ? |
| 11:47.34 | *** join/#asterisk nighty^ (n=nighty@p1022-adsau16honb13-acca.tokyo.ocn.ne.jp) |
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| 11:50.09 | denzs | can anybody give me a hint or a link for using realtime in macros for ael-dialplans? i want to use a macro which is stored in a mysql table...that works great by using the old dialplan syntax, but i cant get it to work with ael :( |
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| 12:16.07 | kamanashisroy | is there any xmlrpc based realtime module ? |
| 12:17.53 | kamanashisroy | if not then is it good idea to write one ? |
| 12:19.25 | jfg | kamanashisroy: did you try AJAM ? |
| 12:20.08 | mvanbaak | kamanashisroy: there's res_config_curl |
| 12:20.14 | kamanashisroy | actually my database is too much fragmented .. writing view needs too many joins .. I have seen that most of the time asterisk search for one entity than numerous ones .. I mean we can easily avoid too much table join .. we can use the important keys to search an entity and get details to get performance benefit .. |
| 12:21.40 | kamanashisroy | jfg: following .. |
| 12:21.47 | kamanashisroy | mvanbaak: following .. |
| 12:22.53 | mvanbaak | you have to write something that returns data when you visit a webpage |
| 12:23.00 | mvanbaak | it will be fetched with curl |
| 12:23.06 | mvanbaak | and you can use that as realtime config |
| 12:23.19 | mvanbaak | I haven't looked at it myself |
| 12:23.28 | kamanashisroy | mvanbaak: and we need to send parameters as urlencoded data .. right ? |
| 12:23.39 | mvanbaak | yup |
| 12:24.13 | mvanbaak | but I think res_config_curl does that |
| 12:24.17 | kamanashisroy | mvanbaak: I mean that addon will send urlencoded data and I shall reply with something formatted .. OK ..following .. |
| 12:24.24 | mvanbaak | yup yup |
| 12:25.02 | mvanbaak | if you want more info, I think Corydon76-dig can help you with it. or give you pointers to the right documentation |
| 12:25.49 | kamanashisroy | jfg: it does not seem like realtime module rather it is like manager .. |
| 12:26.05 | kamanashisroy | mvanbaak: thank you .. I shall follow the code .. |
| 12:27.12 | kamanashisroy | mvanbaak: it has a drawback that it needs to create new http connection each time asterisk want to know something :( |
| 12:28.11 | kamanashisroy | mvanbaak: do you think writing stored procedure based realtime module is better ? |
| 12:29.14 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:31.35 | jfg | kamanashisroy: sorry i misunterstood your phrase |
| 12:31.49 | jfg | ajam is not what you are looking for |
| 12:32.18 | kamanashisroy | jfg: not at all .. |
| 12:32.27 | jfg | sorry :) |
| 12:32.29 | kamanashisroy | jfg: thank you for your help .. |
| 12:32.43 | kamanashisroy | jfg: but ^^ what do you think about that ? |
| 12:34.00 | kamanashisroy | jfg: do not be sorry (I know I have a problem that I cannot ask question clearly :( ) .. do you think stored procedure based realtime module is better ? |
| 12:34.42 | kamanashisroy | it will enable little more intelligence than writing blind view .. |
| 12:34.59 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 12:36.58 | mvanbaak | no idea |
| 12:37.06 | mvanbaak | I'm not using realtime |
| 12:37.16 | mvanbaak | I use my own tool to create config files |
| 12:37.27 | *** join/#asterisk Chris-NB (n=chris@213162066159.public.t-mobile.at) |
| 12:37.30 | jfg | i'm using realtime with sql or ldap and that's all |
| 12:37.30 | kamanashisroy | mvanbaak: OK .. thanks |
| 12:38.10 | kamanashisroy | jfg: OK .. sql realtimes just needs too many joins :( |
| 12:39.15 | marc7 | hey guys... if ControlPlayback dies because the call ends... is there any way I can avoid having it throw warnings about trying to jump to priority n+101? |
| 12:40.43 | *** join/#asterisk pdugas (n=pdugas@74.95.28.33) |
| 12:41.08 | jfg | kamanashisroy: i try to understand your problem and your xmlrpc solution, but i'm busy |
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| 12:44.57 | Uatec | hey there |
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| 13:00.00 | Greek-Boy | I have setup a IAX2 trunk between two * boxes and something strange is happening. I can call from Box A to Box B but when I try to place a call from Box B to Box A it gives me an error. |
| 13:00.07 | Greek-Boy | IAX2 debug shows: CAUSE : No such context/extension |
| 13:00.18 | Greek-Boy | CAUSE CODE : 3 |
| 13:00.26 | Katty | yawns |
| 13:00.32 | Katty | morning |
| 13:00.43 | Greek-Boy | even though the context does exist... |
| 13:00.54 | marc7 | Greek-Boy: what does the iax.conf on Box A look like? |
| 13:00.55 | Greek-Boy | Katty: Good morning |
| 13:01.06 | marc7 | aaand... |
| 13:01.34 | marc7 | can you paste that, and the Dial() cmd you're using... into a pastebin? |
| 13:01.45 | Greek-Boy | alright |
| 13:01.57 | Katty | hello, Greek-Boy |
| 13:03.23 | Corydon76-dig | kamanashisroy: res_config_curl was written to serve as an intermediary for databases which have poor ODBC support |
| 13:03.57 | Corydon76-dig | kamanashisroy: there's a backend for it, written in Perl, in contrib/scripts/dbsep.cgi |
| 13:04.25 | kamanashisroy | Corydon76-dig: thanks for information .. |
| 13:04.43 | Greek-Boy | marc7: iax.conf http://pastebin.com/d281a5865 |
| 13:04.57 | kamanashisroy | Corydon76-dig: it has a drawback that it needs to create new connection whenever it needs to know something .. |
| 13:05.20 | kamanashisroy | Corydon76-dig: ^^ feels bad for performance .. |
| 13:05.50 | Corydon76-dig | kamanashisroy: it massively improved stability for a guy who was using Progress DB |
| 13:05.53 | kamanashisroy | Corydon76-dig: but it serves the aim ... |
| 13:06.24 | kamanashisroy | Corydon76-dig: it is good for bad database engines .. |
| 13:06.48 | Corydon76-dig | Not bad database engines, just databases with poor ODBC connectors |
| 13:06.51 | kamanashisroy | Corydon76-dig: do you think stored procedure based realtime engine is good ? |
| 13:07.37 | kamanashisroy | Corydon76-dig: ^^ I admit .. I was mistaken when I said bad database engine .. |
| 13:07.59 | Corydon76-dig | You're welcome to try |
| 13:08.14 | Katty | what does everyone use to pull up their linux gui on their windows machines? |
| 13:08.17 | [TK]D-Fender | Greek-Boy: 235-240. WTF? You can't assign multiple contexts to a peer. |
| 13:08.21 | kamanashisroy | Corydon76-dig: thanks .. |
| 13:09.01 | marc7 | Greek-Boy: just confirming, in that configuration, 'internal' is the default context for the [boxb] friend, as it's the first one listed... and that context does exist on boxa, right? |
| 13:09.24 | marc7 | Greek-Boy: then, what's the dial command you're using on boxb? |
| 13:09.30 | Corydon76-dig | Katty: who uses Windows, anyway? |
| 13:09.32 | Greek-Boy | marc7; thats correct |
| 13:09.39 | Katty | Corydon76-dig: me. |
| 13:09.44 | Katty | Corydon76-dig: obviously (= |
| 13:09.45 | Corydon76-dig | Katty: try Firefox |
| 13:09.53 | Katty | blinks |
| 13:10.14 | Katty | last i checked, firefox does not do anything like i'm asking for |
| 13:10.49 | Darthclue | you mean the linux command line? or a gui as in x-windows? |
| 13:11.08 | Katty | what does everyone use to pull up their linux gui on their windows machines? <- what i typed. |
| 13:11.15 | Greek-Boy | [TK]D-Fender; its something i forgot to fix |
| 13:11.16 | jameswf-home | doesn't have a linux machine.... |
| 13:11.25 | Corydon76-dig | Katty: you mean the Asterisk GUI? |
| 13:11.31 | Katty | no. |
| 13:11.34 | Katty | my asterisk does not have a gui |
| 13:11.37 | Darthclue | hasn't used a linux gui in ages |
| 13:11.40 | Sir0x | i am forced to use windows at work :( |
| 13:12.00 | [TK]D-Fender | Greek-Boy: And you're wondering why you can't find extens in the right context.... you should work on the stuff in the big print first.... |
| 13:12.01 | [TK]D-Fender | Katty: Mew. |
| 13:12.02 | Katty | i can see this question is hopeless.i will ask elsewhere. |
| 13:12.08 | Corydon76-dig | Katty: I don't understand what you're trying to do |
| 13:12.09 | Katty | [TK]D-Fender: hi. |
| 13:12.13 | Darthclue | katty, a vnc client would do it if you setup a vnc server on the linux box |
| 13:12.21 | Greek-Boy | marc7: exten => _0797120XXX,1,Dial(IAX2/boxa/${EXTEN:7}) |
| 13:12.30 | jameswf-home | my company is pretty much linux and osx except the accountants.... as soon an ineuit gets off their asses though |
| 13:12.57 | jameswf-home | *intuit |
| 13:13.14 | Greek-Boy | [TK]D-Fender: I thought you can assign multiple contexts and that the first one is the default? |
| 13:13.31 | [TK]D-Fender | Greek-Boy: No. ! context period like every other device. |
| 13:13.34 | [TK]D-Fender | 1* |
| 13:13.47 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 13:13.55 | jameswf-home | Greek-Boy: you can include => context |
| 13:13.55 | marc7 | Greek-Boy: so dialing 0797120123 should theoretically be looking for 123 in the [internal] context on boxa, does that exist? |
| 13:13.57 | Greek-Boy | [TK]D-Fender: ok i got it. I will make use of the include command. |
| 13:14.09 | _ShrikE | Katty: winaxe |
| 13:14.15 | Greek-Boy | marc7 yes it does |
| 13:14.18 | *** join/#asterisk LaSombra (n=lasombra@mvx-200-201-179-34.mundivox.com) |
| 13:14.19 | Katty | _ShrikE: is that server or client? |
| 13:14.24 | LaSombra | Morning |
| 13:14.36 | _ShrikE | client |
| 13:14.46 | jameswf-home | thinks windoze is over raed and has been oure linux over 2 years.... |
| 13:14.47 | Katty | _ShrikE: what server does it use to connect to? |
| 13:15.08 | Greek-Boy | marc: I am trying to dial 0797120712 which is in the extensions.conf on http://pastebin.com/d4d65ad8d |
| 13:15.12 | _ShrikE | It uses xwindws |
| 13:15.16 | Corydon76-dig | Katty: which GUI is running on the Linux box? |
| 13:15.41 | LaSombra | Question: What's the difference between chan_oh323 and chan_h323? |
| 13:16.02 | Corydon76-dig | LaSombra: I give up |
| 13:16.12 | jameswf-home | LaSombra: an "o" |
| 13:16.13 | Katty | _ShrikE: hrm. no sign of that in apt-cache |
| 13:16.14 | *** join/#asterisk coppice (n=chatzill@99.166.17.210.dyn.pacific.net.hk) |
| 13:16.16 | Corydon76-dig | LaSombra: what's the punchline? |
| 13:16.25 | Katty | _ShrikE: does it have another name by chance? (= |
| 13:16.43 | _ShrikE | http://www.winaxe.com/xwindows-for-windows.html |
| 13:16.53 | LaSombra | Corydon76-dig: I need to integrate with an Avaya PBX and use Avaya phones. |
| 13:16.56 | Katty | oh. xwindows runs on windows? That wont' work. |
| 13:16.57 | jameswf-home | Katty X is the server on top of which all guis are built. |
| 13:17.08 | marc7 | [TK]D-Fender: hey, i might learn a thing or two from Greek-Boy's question. you said that you can't assign multiple contexts to an IAX2 peer? does that only work for IAX2 users? |
| 13:17.09 | *** join/#asterisk JayTee52 (n=jforde@207-67-84-177.static.twtelecom.net) |
| 13:17.12 | Katty | jameswf-home: i am not stupid. |
| 13:17.24 | Corydon76-dig | LaSombra: If you value your sanity, you'll use SIP |
| 13:17.24 | jameswf-home | whistles |
| 13:17.25 | [TK]D-Fender | marc7: on AND device |
| 13:17.27 | LaSombra | Corydon76-dig: But I read a lot of people is using oh323, but that doesn't make sense on a 1st look since h323 is "embedded" |
| 13:17.28 | [TK]D-Fender | ANY* |
| 13:17.41 | LaSombra | Corydon76-dig: Can't. All my phones are h323 :( |
| 13:18.03 | marc7 | [TK]D-Fender: I'm misreading http://www.voip-info.org/wiki-Asterisk+config+iax.conf which says near the top that "Multiple permitted contexts may be specified, in which case the first will be the default." |
| 13:18.06 | [TK]D-Fender | Katty: What are you trying to do, use your Linux desktop on a Windows PC? |
| 13:18.08 | jameswf-home | sanity is overrated :) |
| 13:18.08 | _ShrikE | Katty: the tutorial on their website will walk you through it. |
| 13:18.12 | marc7 | sorry |
| 13:18.18 | Katty | [TK]D-Fender: yes. |
| 13:18.34 | jameswf-home | KDE4 runs on windoze... so I hear |
| 13:18.37 | Katty | [TK]D-Fender: like vnc, or something similiar. |
| 13:19.11 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 13:19.14 | Corydon76-dig | Katty: there's a builtin VNC server for KDE, but you have to enable it (disabled by default) |
| 13:19.32 | [TK]D-Fender | marc7: Indeed poorly worded. they mean SECTION names like [john] can appear multiple times with different permit/deny, etc. |
| 13:19.41 | Katty | Corydon76-dig: i dont' like kde. i usually use gnome. |
| 13:19.42 | Greek-Boy | marc7: i corrected what [TK]D-Fender pointed out. I now have only one context listed for that friend but the same problem and cause code persists. |
| 13:19.54 | [TK]D-Fender | Katty: http://sourceforge.net/projects/xming |
| 13:20.37 | jameswf-home | uses KDE unless its embeded like a thin client then uses windowmaker |
| 13:20.45 | LaSombra | Corydon76-dig: That's why I need to understand the difference between both versions. I can't find it on google :( |
| 13:20.54 | Katty | [TK]D-Fender: this says it's for windows |
| 13:21.05 | Katty | [TK]D-Fender: if this is for windows, what server backend does it connect to? |
| 13:21.06 | marc7 | [TK]D-Fender: noted! starting to do more iax-y stuff these days so that'll help. // Greek-Boy: I'm still looking at it, but I'm not very fast at picking out problems. Others here may be able to help too. |
| 13:21.19 | Corydon76-dig | LaSombra: the main difference is licensing |
| 13:21.30 | LaSombra | Corydon76-dig: Not features? |
| 13:21.34 | [TK]D-Fender | LaSombra: You're pretty screwed. * doesn't handle H.323 well. |
| 13:21.42 | Corydon76-dig | Nope, they both use exactly the same library |
| 13:22.13 | jameswf-home | flips off the light switch in a room with no windows and asks "Why the hell is it dark in here?" |
| 13:22.15 | LaSombra | [TK]D-Fender: That's not very encouraging.. heh :) |
| 13:22.19 | LaSombra | Corydon76-dig: Ok. thanks :) |
| 13:22.28 | [TK]D-Fender | LaSombra: How big is your Avaya setup? |
| 13:22.33 | Corydon76-dig | [TK]D-Fender: openh323 doesn't handle h.323 well |
| 13:22.45 | Corydon76-dig | [TK]D-Fender: which is the problem |
| 13:23.16 | LaSombra | [TK]D-Fender: 1 call center with 600+ agents and about 7 offices with 100+ ppl each |
| 13:23.42 | Corydon76-dig | LaSombra: and why are you integrating Asterisk again? |
| 13:24.27 | LaSombra | Corydon76-dig: For a new small office with 15 ppl where the company that handles our Avayas are charging us about 15k |
| 13:24.56 | Corydon76-dig | I thought the Avaya PBX let you connect with SIP |
| 13:25.00 | LaSombra | Corydon76-dig: Also it'll be our pilot project for changing our current Avaya installation. Step by step |
| 13:25.17 | LaSombra | Corydon76-dig: Could be, but we have H323 physical phones that we want to use |
| 13:26.05 | Corydon76-dig | LaSombra: you're going to spend $15k trying to figure out how to get those phones to work with Asterisk... and then you'll spend another $5k for SIP phones. |
| 13:26.16 | Corydon76-dig | Save the money. Just buy SIP phones up-front |
| 13:26.48 | Corydon76-dig | Or check with Avaya if those phones have SIP images |
| 13:27.00 | hmmhesays | That they do |
| 13:27.03 | LaSombra | Corydon76-dig: We already have the phone laying around. That's why. Avaya wants to charge for the 15k for the PBX and installation services |
| 13:27.19 | LaSombra | Corydon76-dig: Hmmmm, SIP images... great... didn't think of that |
| 13:27.29 | hmmhesays | 15K? holy crap |
| 13:27.50 | LaSombra | hmmhesays: Yeah, that's why I will work my ass off to make this work |
| 13:37.55 | sysreq` | ~thebook |
| 13:37.55 | 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 |
| 13:38.07 | *** join/#asterisk RoyK (n=roy@fw.fortel.no) |
| 13:38.13 | *** join/#asterisk jm|home (n=jm|home@dilbert.jamiem.com) |
| 13:38.14 | sysreq` | sorry, needed the link ;p |
| 13:38.24 | *** join/#asterisk af_ (n=getsmart@88-149-240-26.dynamic.ngi.it) |
| 13:39.42 | *** join/#asterisk d3wayne (n=deeewayn@76.29.245.9) |
| 13:39.42 | *** mode/#asterisk [+o d3wayne] by ChanServ |
| 13:40.08 | [TK]D-Fender | LaSombra: What model of phones are you looking to reuse? |
| 13:41.35 | *** join/#asterisk lftsy (n=lftsy@121.194.210.62.te-dns.org) |
| 13:44.11 | LaSombra | [TK]D-Fender: Avaya 4610SW IP |
| 13:45.20 | [TK]D-Fender | LaSombra: Yes, they appear to support SIP |
| 13:45.55 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:45.55 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:46.24 | LaSombra | [TK]D-Fender: Hmmmm, I'll take a look with my distributor. Thanks man |
| 13:54.29 | *** join/#asterisk eric_hill (i=EricHill@204.94.175.11) |
| 13:56.10 | mocker | Great, I upgrade my sip.ld on my polycom and now my buddy list/presence isn't working. |
| 13:56.14 | mocker | :| |
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| 14:01.48 | *** join/#asterisk ACiDV (n=joel@122-205-229.dr.cgocable.ca) |
| 14:01.55 | kamanashisroy | hi, I have not used queue .. unfortunately I am writing a gui for queue for support team .. anyway I have created table for queue member in realtime .. I cannot sort out how to use 'interface' .. I mean how the call is terminated through interface ? can interface take a value like sip/someagent@gateway ? |
| 14:03.09 | ACiDV | Anyone using Zaptel in Saudi Arabia (Dubia, Jubara, etc) ? Have problem w/ indications and since no toneindications defined on asterisk for S.A., not sure which to use, if one can work of if I must "write" it (using telco specs..) |
| 14:04.02 | *** join/#asterisk javar (n=javar@69.79.134.24) |
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| 14:11.07 | *** join/#asterisk UnixDog (n=UnixDog@adsl-69-230-162-152.dsl.irvnca.pacbell.net) |
| 14:11.12 | *** join/#asterisk wonderworld (n=voici@ip-62-143-163-248.1211G-CUD12K-01.ish.de) |
| 14:11.50 | wonderworld | hi, after rebooting my box, asterisk stutters extremely when doing Playback()s. I didn't change a thing, just rebooted. What may be the cause? tnx |
| 14:13.16 | Corydon76-dig | wonderworld: is zaptel started? |
| 14:13.21 | UnixDog | wonderworld: need more info: os, mem, cpu speed, network status |
| 14:13.38 | UnixDog | zaptel status |
| 14:14.16 | wonderworld | Corydon76-dig: i am not using zaptel, dialing in with SIP |
| 14:14.23 | rickross | anyone know of a problem with 3-way conferencing using Polycom phones with Asterisk - SIP version 3.0.0? |
| 14:14.23 | UnixDog | how many times you have wacked it with a rubber malet |
| 14:14.30 | wonderworld | UnixDog: that shouldn't be a matter. it worked just fine for days on the same box |
| 14:14.37 | wonderworld | the reboot must have changed something |
| 14:14.39 | wonderworld | it's debian |
| 14:14.49 | Corydon76-dig | wonderworld: doesn't matter. If you were running zaptel before, you need to be, still |
| 14:14.55 | UnixDog | rickross: 3.0.0 is a issue |
| 14:15.02 | UnixDog | backsetp to 2.2.2 |
| 14:15.04 | wonderworld | ok, let me check |
| 14:15.09 | rickross | our users who have 2.x firmware on the Polycom phones have no problem, but the 3.0.0 Polycom firmware users get a fast-busy signal when they try to conference in a second call |
| 14:15.11 | Corydon76-dig | wonderworld: SIP uses the zap timing device if it was compiled |
| 14:15.27 | wonderworld | goldbin*CLI> zaptel status |
| 14:15.27 | wonderworld | No such command 'zaptel' (type 'help' for help) |
| 14:15.53 | Corydon76-dig | wonderworld: lsmod | grep zap |
| 14:15.56 | UnixDog | wonder at the cli zap show channels |
| 14:15.59 | rickross | UnixDog - are you aware of any further details about this 3.0.0 conference problem? |
| 14:16.02 | UnixDog | not zaptel |
| 14:16.20 | rickross | is this an Asterisk issue, a Polycom issue? Can we help debug it? |
| 14:16.27 | wonderworld | goldbin*CLI> zap show channels |
| 14:16.27 | wonderworld | <PROTECTED> |
| 14:16.27 | wonderworld | <PROTECTED> |
| 14:16.38 | UnixDog | 3.0.0 current know to corrupt bootroms and many othe feature of the polycom |
| 14:16.48 | wonderworld | lsmod | grep zap -> nothing |
| 14:16.59 | Corydon76-dig | wonderworld: modprobe ztdummy |
| 14:17.01 | UnixDog | lcr speeddial all fail to work |
| 14:17.20 | wonderworld | FATAL: Module ztdummy not found. |
| 14:17.28 | Corydon76-dig | wonderworld: that's the problem |
| 14:17.31 | wonderworld | hmm, strange :) |
| 14:17.35 | rickross | UnixDog: our bootroms are updated to the latest from Polycom - the phones are mostly working well, but the 3-way conferencing is failing |
| 14:17.59 | rickross | UnixDog: any sources for details about the problems? We haven't been able to find anything useful yet |
| 14:18.01 | Corydon76-dig | wonderworld: did you recently upgrade your kernel? |
| 14:18.11 | wonderworld | nope |
| 14:18.13 | UnixDog | I have personally seen the 3.0.0 currupt 15 polycoms with bootrom 4.0 |
| 14:18.19 | Corydon76-dig | wonderworld: reinstall zaptel |
| 14:18.32 | wonderworld | hmm i just installed the debian asterisk packages |
| 14:18.55 | UnixDog | let me grap and test a phone for the 3way issue. |
| 14:18.56 | Corydon76-dig | Oh, in that case, you need to contact the Debian maintainer with questions |
| 14:19.06 | wonderworld | hehe, thanks anyway |
| 14:19.08 | UnixDog | I have one if it bricks I dont care |
| 14:19.10 | Corydon76-dig | This the support channel for people who compile from source |
| 14:19.12 | rickross | UnixDog: much appreciated |
| 14:19.17 | UnixDog | it has a dying screen |
| 14:19.39 | rickross | UnixDog: we're using Polycom SoundPoint IP 550's, in case that matters |
| 14:20.04 | UnixDog | I have a 550 a 600 a 501 and a 320 |
| 14:20.20 | UnixDog | 550's rock |
| 14:20.21 | UnixDog | lol |
| 14:21.02 | JayTee52 | I never trust Asterisk in .deb or .rpm. |
| 14:21.08 | UnixDog | ok flashing the 501 now |
| 14:21.24 | UnixDog | does not trust asterisk on linux |
| 14:21.30 | UnixDog | ducking |
| 14:21.40 | wonderworld | well, i started trusting it |
| 14:21.53 | UnixDog | not wanting to be pelted with linux admin manuals |
| 14:22.24 | JayTee52 | lemme guess, running it on BSD? |
| 14:22.56 | UnixDog | yeserie |
| 14:23.03 | *** join/#asterisk Sniper_lilili (n=michofr@mail.splendor.net) |
| 14:23.09 | wonderworld | i fixed it |
| 14:23.18 | wonderworld | i better won't tell how...... |
| 14:23.27 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 14:23.31 | wonderworld | you guys would have an even worse oppinion on me ;) |
| 14:23.36 | JayTee52 | ok, if I was gonna stereotype ya I'd bet ya got a ponytail, wear flannel shirts and jeans, have a soul patch on your chin and an earring. |
| 14:23.37 | Sniper_lilili | Hi all...Can you please let me know where I can Install G729 codec for my asterisk server? and what is the file name? |
| 14:23.48 | JayTee52 | and vote either liberal democrat or green party |
| 14:23.57 | UnixDog | JayTee52: nope you got it wrong |
| 14:24.01 | Sniper_lilili | I want it as free please |
| 14:24.09 | JayTee52 | it's why i avoid stereotyping |
| 14:24.28 | JayTee52 | and gambling in Vegas |
| 14:24.38 | UnixDog | I am 40 I wear denum shorts and bigdog polo shirts and boots and have a flattop |
| 14:25.06 | JayTee52 | ah, the new Unix warrior |
| 14:25.25 | JayTee52 | I can't wear jeans or shorts at work. |
| 14:25.37 | JayTee52 | only the keepers can wear shorts in the summer |
| 14:25.40 | UnixDog | some times I wear a utility kilt that has a bunch of logos on it |
| 14:25.54 | JayTee52 | I.T. has to wear casual dress |
| 14:26.17 | UnixDog | freebsd/kde/asterisk/sangoma/and a few other logos |
| 14:27.11 | Sniper_lilili | Hi all...Can you please let me know where I can Install G729 codec for my asterisk server? and what is the file name? |
| 14:27.22 | [TK]D-Fender | Sniper_lilili: www.digium.com |
| 14:27.28 | JayTee52 | when did the first release of freeBSD come out? |
| 14:27.36 | [TK]D-Fender | Sniper_lilili: Downloads & documentation can be found there |
| 14:27.58 | Sniper_lilili | [TK]D-Fender, I need something for free |
| 14:28.15 | [TK]D-Fender | Sniper_lilili: G.729 is a licensed codec, not a free one. |
| 14:28.45 | JayTee52 | what codec do most local telcos use in the states for PRI circuits? |
| 14:28.47 | Sniper_lilili | [TK]D-Fender, do you know any web site where I can download a free one for testing purpose? |
| 14:29.07 | JayTee52 | or is it just for the phones? |
| 14:29.09 | [TK]D-Fender | Sniper_lilili: Not offhand, and this is not something to be asked here. |
| 14:29.44 | Sniper_lilili | [TK]D-Fender, thanks for help |
| 14:29.44 | UnixDog | i386bsd ver 1.0 came out in 1982 I believe |
| 14:29.48 | *** join/#asterisk Claude2008 (n=Claude81@87-194-221-2.bethere.co.uk) |
| 14:30.55 | JayTee52 | UnixDog, I'm pretty sure I tried 386BSD in 92 or 93. |
| 14:31.51 | UnixDog | I used it in collage |
| 14:32.00 | UnixDog | and that was 21 years ago |
| 14:32.13 | UnixDog | cant speel this am |
| 14:32.16 | *** join/#asterisk NoamRotter (n=noam@mail.browarnik.com) |
| 14:32.18 | UnixDog | lol spell |
| 14:32.29 | UnixDog | need to put my contacts in brb |
| 14:32.43 | *** join/#asterisk jmesquita (n=jmesquit@201.7.117.114) |
| 14:33.28 | UnixDog | he first CDROM (and general net-wide) distribution was FreeBSD 1.0, released in December of 1993. This was based on the 4.3BSD-Lite (âNet/2â) |
| 14:33.35 | UnixDog | he/the |
| 14:33.39 | Claude2008 | hello |
| 14:33.50 | Claude2008 | any one able to help |
| 14:34.12 | Claude2008 | i just installed trixbox 2.6.0.7 |
| 14:34.23 | Claude2008 | and i am having problem getting the zap channles to work |
| 14:35.02 | [TK]D-Fender | ~trixbox |
| 14:35.02 | jbot | [~trixbox] trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org |
| 14:35.10 | *** join/#asterisk jmesquita (n=jmesquit@201.7.117.114) |
| 14:35.41 | Claude2008 | ok thanks |
| 14:41.29 | NoamRotter | any1? |
| 14:43.54 | Nasra | I have not been able to use Asterisk yet ...still learning linux... |
| 14:44.50 | *** join/#asterisk Shotygun (n=thorn@213.31.43.3) |
| 14:46.14 | Shotygun | Hello. Does anybody know how to elimnate make Snom phones not to show "Messages-Wait", the cut-off SIP MWI text and instead showing some more useful indication ? |
| 14:46.17 | UnixDog | get pbxinaflash |
| 14:46.27 | UnixDog | do away with all the extra crap |
| 14:46.35 | UnixDog | better yet get asterisk-now |
| 14:46.45 | UnixDog | the digium developed gui+asterisk |
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| 14:48.05 | *** join/#asterisk ccole (n=cole@cpe-71-72-50-240.neo.res.rr.com) |
| 14:48.30 | ccole | Anyone else have a shared library problem with /usr/lib64/asterisk/modules/chan_iax2.so recently?? |
| 14:49.07 | ccole | Here's the error: asterisk: symbol lookup error: /usr/lib64/asterisk/modules/chan_iax2.so: undefined symbol: ast_qched_runq |
| 14:49.26 | ccole | I ran hexedit on the shared lib, changed qched to sched, and everything works find now :-/ |
| 14:49.31 | ccole | find=fine |
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| 15:07.11 | UnixDog | rickross: ok I have tested with changing settings in the sip.cfg and the phone_mine.cfg and the nway conferencing does not work with 3.0.0. I have contacted Polycom support about this issue . and they said I should have a call back in the next few hours |
| 15:08.17 | rickross | Unix-Dog: thanks - we're trying to do standard 3-way conferencing , not the new N-way (which requires a feature license $$$ I think) |
| 15:08.42 | UnixDog | nway/3way are the same |
| 15:08.51 | UnixDog | and as far as I know it does not |
| 15:08.55 | UnixDog | but neither work |
| 15:09.06 | UnixDog | and I have reported it to the support team |
| 15:09.36 | rickross | UnixDog: N-way comes with some additional management application support - the traditional 3-way is separate, I believe |
| 15:09.50 | rickross | but thanks |
| 15:10.14 | rickross | they probably broke it while setting up to milk us all for $$$ for the "feature activation" |
| 15:10.23 | UnixDog | lol |
| 15:10.25 | UnixDog | true |
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| 15:16.04 | UnixDog | brb |
| 15:16.13 | UnixDog | doing a rip and rebuild on my dsl line |
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| 15:23.19 | coolhp | Good day everyone |
| 15:23.55 | coolhp | Is anyone running Asterisk 1.4.19 having problems with the Background() function ignoring the channel language ? |
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| 15:30.12 | *** join/#asterisk nybbled (n=nybbled@about/apple/performa/nybble) |
| 15:30.17 | nybbled | hello all. |
| 15:32.26 | nybbled | does anyone know that for the chan_iax implementation. Whether or not, if a call were to come into one system, be sent to a second system, and then transferred via a SIP phone to an extension on the originating server, whether or not the call is unhooked from the first server and then routed to that local extension on the first server, or is the call sent back through on a second channel back to the first system |
| 15:34.06 | ManxPower | nybbled: no, that does not happen |
| 15:34.14 | cr0n | hi, i have a digium TE122P card on a E1 line, incoming calls to any one of the numbers works through and goes to the extension specified however, i cannot dial out even though ive set a route to go out the same interface thats already working for all incoming calls, any tips? |
| 15:34.39 | nybbled | ManxPower, so the call is sent back to the first server, on a second channel? |
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| 15:36.23 | [TK]D-Fender | cr0n: Pastebin the complete CLI output of your failed attempt at verbose 10 Include your zaptel.conf, and you zapata.conf |
| 15:36.27 | [TK]D-Fender | ~pb |
| 15:36.28 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 15:36.29 | [TK]D-Fender | ^^^^^^^^^^ |
| 15:37.05 | cr0n | [TK]D-Fender: one moment |
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| 15:38.48 | nybbled | ManxPower: woops, now that I re-read what you typed, it was my second option that you said was what was happening |
| 15:40.48 | cr0n | [TK]D-Fender: http://pastebin.com/m96bf538 |
| 15:43.38 | lirakis_work | did 1.4 change how it looks for multi-language sound files? |
| 15:44.22 | lirakis_work | i have /var/lib/asterisk/sounds/digits/es/ ... with all my spanish digits sounds... but when I set lang=es .. it still uses english digits |
| 15:44.43 | lirakis_work | i wasnt sure if they changed the structure of sound file storage |
| 15:44.49 | [TK]D-Fender | cr0n: -- Executing [s@macro-dialout-trunk:20] Dial("SIP/3000-08c4c038", "ZAP/g0,11/0822322323|300|") in new stack |
| 15:44.57 | [TK]D-Fender | cr0n: this is not a valid vormat. |
| 15:45.14 | nybbled | does anyone have a suggestion for having this type of thing implemented? |
| 15:45.29 | cr0n | [TK]D-Fender: the trunk name/channels wrong or what? |
| 15:45.35 | [TK]D-Fender | cr0n: Fix your dial rule so it does Dial("SIP/3000-08c4c038", "ZAP/g0/somenumberwithoutextracrapinit|300|") |
| 15:46.45 | cr0n | [TK]D-Fender: what extra stuff..the 082xxxx that you see is the dialed number? |
| 15:47.14 | lirakis_work | nybbled: that depends on if you have media going through each server.. but the signaling will be chained .. the far end, only knows its closest endpoint for signalling .. eg the sip phone only knows the server that sent it the call .. not the originating server. |
| 15:47.27 | lirakis_work | nybbled: unless you use stateless messaging (like redirects) |
| 15:47.34 | [TK]D-Fender | cr0n: "ZAP/g0,11/0822322323|300|" < the "," after "g0", the extra "/" after "11" |
| 15:47.37 | [TK]D-Fender | cr0n: both bad. |
| 15:48.00 | lirakis_work | nybbled: then the call doesnt setup till it reaches the destination, and messaging will flow between the two endpoints and cut out all the intermediate servers |
| 15:49.31 | cr0n | [TK]D-Fender: got it down to: -- Executing [s@macro-dialout-trunk:20] Dial("SIP/3000-08c8a818", "ZAP/g0/0822122120|300|") in new stack |
| 15:49.41 | cr0n | extra / is still there though |
| 15:49.55 | [TK]D-Fender | cr0n: better. Now is "0822122120" a valid # to dial against your PRI? |
| 15:50.05 | [TK]D-Fender | cr0n: No, that line looks fine. |
| 15:50.43 | cr0n | [TK]D-Fender: its a valid working number, yes. |
| 15:50.50 | nybbled | lirakis_work: hmm. Ok. Well, on the the far side server (the one getting the forwarded calls), they are transferring to an extension entry to ( Dial(IAX2/otherserver/extension) ) |
| 15:50.54 | [TK]D-Fender | cr0n: Then it looks fine. |
| 15:51.38 | cr0n | [TK]D-Fender: still not working, should i update the pastebin with the new CLI output? |
| 15:51.50 | lirakis_work | nybbled: your original "question" doesnt ask anything specific... what is it you want to accomplish |
| 15:52.30 | [TK]D-Fender | cr0n: Sure, and pastbin zapata.conf again as well as zapata.conf and every file it links to |
| 15:52.56 | cr0n | [TK]D-Fender: its included in the original post, i changed nothing since the first post. |
| 15:53.40 | lirakis_work | so.. i saw "(NOTE: The structure for 1.4 is diferent but it requires a configuration change) " regarding the sound file structure .. but it gives no info about the new changes.. anyone know where this is documented? |
| 15:54.41 | [TK]D-Fender | cr0n: No, zapata.conf includes other files. |
| 15:54.48 | [TK]D-Fender | cr0n: I need to see those as well. |
| 15:54.53 | [TK]D-Fender | cr0n: all of them. |
| 15:55.07 | cr0n | [TK]D-Fender: one sec. |
| 15:55.27 | nybbled | lirakis_work: my apologies. I've got a receptionist on the second server, who is answering for the first server. when she is sending calls back to the first server, this takes two channels, and more bandwidth then we can afford. We'd like to find a way to use only one channel for this (pretty much have the second server send back to the first one where to put the call, and have the first server move it locally, thus not taking up the second |
| 15:56.31 | lirakis_work | nybbled: are these servers in the same geographic location, or is it possible to have a private lan between them? |
| 15:57.10 | nybbled | in two cities |
| 15:57.40 | lirakis_work | nybbled: call -> server2-> receptionist transfer -> server1 |
| 15:57.41 | nybbled | lirakis_work: i wish i could just physically link them |
| 15:57.53 | jer | get permits to dig and lay a fibre cross connect =] |
| 15:57.54 | *** join/#asterisk qdk (n=qdk@195.242.194.42) |
| 15:58.17 | lirakis_work | nybbled: setup a trunk between the two servers, enable reinvites so that media can traverse between endpoints, and not through the whole call path |
| 15:58.44 | nybbled | reinvites on SIP and IAX? |
| 15:58.50 | cr0n | [TK]D-Fender: http://pastebin.ca/977020 |
| 15:58.58 | lirakis_work | nybbled: on the end users of sip |
| 15:59.04 | nybbled | ok, |
| 15:59.07 | lirakis_work | nybbled: calls come in sip ? or no |
| 15:59.30 | nybbled | lirakis_work: oh sorry... |
| 15:59.37 | lirakis_work | nybbled: you want to reivinte the media between the end caller .. and the originating point |
| 16:00.09 | nybbled | lirakis_work: call (on server 1) -> (iax tunnel) -> receptionist (server 2) -> extension (server 1) |
| 16:00.10 | lirakis_work | nybbled: either that or setup a proxy so that you can actually route calls |
| 16:00.36 | *** join/#asterisk aiurea (n=aiurea@83.166.220.142) |
| 16:00.56 | aiurea | hi |
| 16:01.17 | nybbled | lirakis_work: the transport between the two sites is IAX2 |
| 16:01.35 | aiurea | what could cause this error: SIP/2.0 400 Bad Request - 'Invalid SDP information'? could it be a codec issue? |
| 16:02.17 | aiurea | I am trying to establish a connection to a fixed phone. It rings, and when I pick up the connection is dropped |
| 16:02.26 | aiurea | and I get that error |
| 16:02.32 | aiurea | googling didn't turn out anything interesting |
| 16:02.38 | [TK]D-Fender | cr0n: try dialing out with decending channel selection "G0" instead of "g0" |
| 16:03.08 | lirakis_work | nybbled: i believe that IAX does allow you to disengage media flows now.. but im not sure how .. (i dont use it) .. but that will free up the bandwidth accross 1 server and the tunnel |
| 16:03.11 | *** join/#asterisk shtoom (n=shtoom@121.246.167.147) |
| 16:03.51 | cr0n | [TK]D-Fender: -- Executing [s@macro-dialout-trunk:20] Dial("SIP/3000-08c8a818", "ZAP/G0/0822122120|300|") in new stack - same error im afraid |
| 16:04.30 | lirakis_work | man im hungry |
| 16:05.48 | ManxPower | cr0n: replace the | with a , |
| 16:06.23 | ManxPower | " -- Channel 0/1, span 1 got hangup, cause 44" is your problem. |
| 16:06.42 | [TK]D-Fender | ManxPower: Said the channel was in use. |
| 16:07.00 | [TK]D-Fender | cr0n: Your PRI dialplan may be the issue |
| 16:07.14 | nybbled | lirakis_work: hmmm. yes, i figured something might exist like that. I'm not sure myself on how to do it. We were thinking we could possibly do a SIP trunk, as long as my thoughts are right: |
| 16:07.15 | cr0n | [TK]D-Fender: ive set it to allow any 10 digit number |
| 16:07.25 | ManxPower | No, Cause Code 44 is Requiested Circuit/Channel not available. |
| 16:07.33 | [TK]D-Fender | cr0n: this has nothing to do with what you dial... its a signalling issue |
| 16:08.03 | ManxPower | BTW, why are we helping people with AMP/FreePBX/Trixbox config files? |
| 16:08.25 | nybbled | lirakis_work: call (server 1) -> (sip trunk) -> answered (server 2) -> transfered via same sip trunk -> extension (server 1) |
| 16:08.44 | nybbled | lirakis_work: media stream would be connected first to remote server then back to originating server? |
| 16:08.49 | nybbled | lirakis_work: using canreinvite=yes |
| 16:08.50 | [TK]D-Fender | ManxPower: I do the minimum if they provides stuff upfront before mentioning what they've got. So long as the answer is easier than debugging crap I'll still try |
| 16:09.02 | ManxPower | nybbled: there is a good chance media will bypass Asterisk in that setup if everything is set up correctly and there is no NAT involved |
| 16:09.16 | [TK]D-Fender | ManxPower: the second they ask how to use the GUI to do it, then its TFB territory :) |
| 16:09.41 | nybbled | ManxPower: yea, we have the choice of using a VPN or straight internet |
| 16:09.52 | ManxPower | [TK]D-Fender: cr0n does seem surprisingly technical for being an AMP user, |
| 16:10.28 | [TK]D-Fender | ManxPower: Wouldn't go that far yet, but not a real twit so far so I'll run with it a little. |
| 16:10.29 | ManxPower | nybbled: just remember it's only AUDIO that gets reinvited, not SIP signaling. |
| 16:10.30 | cr0n | ManxPower: im trying all systems, this just happens to be one of those boxes im afraid. |
| 16:10.51 | ManxPower | cr0n: what is your pridialplan= set to? |
| 16:11.23 | cr0n | and this is also one of the first times im working with pri's - so do excuse me. |
| 16:11.27 | nybbled | ManxPower: yea, the audio is the brunt of the bandwidth use, if we have to deal with having signalling still taking the extra steps, its not much of an issue |
| 16:11.38 | lirakis_work | nybbled: no .. media would go between first server and the end user |
| 16:12.01 | lirakis_work | nybbled: there is a pictorial example of this in thebook |
| 16:12.03 | ManxPower | lirakis_work: assuming the same codec, no DTMF based commands, no recording. |
| 16:12.03 | lirakis_work | ~thebook |
| 16:12.04 | 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 |
| 16:12.18 | *** join/#asterisk MmixX (i=mmixx@202.124.138.69) |
| 16:12.23 | lirakis_work | ManxPower: yes.. i mean * cant be doing anything in the media path |
| 16:12.51 | nybbled | lirakis_work, ManxPower: correct. Media path is going from first server to whever it is been transferred |
| 16:13.04 | *** join/#asterisk _mwoodj_ (n=MWoodJ@pdpc/sponsor/digium/hyper-eye) |
| 16:13.08 | nybbled | lirakis_work: whereis the signalling is still traveling through all the points along the way |
| 16:14.11 | lirakis_work | nybbled: yes |
| 16:14.12 | lirakis_work | nybbled |
| 16:14.15 | lirakis_work | oops |
| 16:14.21 | lirakis_work | nybbled: page 189 of |
| 16:14.23 | lirakis_work | ~thebook |
| 16:14.24 | 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 |
| 16:14.31 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177582637.dsl.bell.ca) |
| 16:14.37 | nybbled | lirakis_work: ok, i got the pdf around here somewhere already |
| 16:15.43 | nybbled | lirakis_work: page 189? |
| 16:15.49 | nybbled | lirakis_work AstDB? |
| 16:15.54 | lirakis_work | nybbled: in dead tree format |
| 16:16.03 | lirakis_work | nybbled: make sure you have the 2nd edition |
| 16:16.32 | lirakis_work | nybbled: its in the understanding telephony chapter |
| 16:16.40 | nybbled | lirakis_work: i've got the second edition pdf, first edition deadtree |
| 16:16.46 | *** join/#asterisk shinao1 (n=shinao1@41.221.165.43) |
| 16:17.10 | lirakis_work | nybbled: i dont have the pdf .. just 2nd DT ed. .. you should be able to find it from here |
| 16:17.40 | nybbled | lirakis_work ok |
| 16:18.29 | cr0n | ManxPower: there was no pridialplan= so ive added pridialplan=local and priindication=outofband in /etc/asterisk/zapata.conf |
| 16:18.48 | ManxPower | cr0n: no, you set it to unknown |
| 16:18.58 | ManxPower | If you set it to local then you will only be able to make local calls |
| 16:19.20 | ManxPower | cr0n: you do not want to set priindication= |
| 16:19.52 | nybbled | i found "Figure 8-1. The SIP trapezoid" |
| 16:20.11 | cr0n | ManxPower: ok, so i should set pridialplan= to? (to allow any type of call) |
| 16:20.21 | nybbled | lirakis_work: yes, signalling runs through all points, rtp unless told otherwise will try to connected directly to the endpoint |
| 16:20.21 | ManxPower | pridialplan=unknown |
| 16:20.34 | cr0n | ManxPower: alright, let me try. |
| 16:20.46 | ManxPower | you missed the note in the default zapata.conf about pridialplan settings? |
| 16:21.15 | cr0n | ManxPower: i guess these prebuild systems gladly remove it for you, as it was not there. |
| 16:22.02 | cr0n | ManxPower: ive just made that change and removed 'priindication=outofband' - restarted, and still all-circuits-are-busy. should i update pastebin with a new output? |
| 16:23.55 | ManxPower | cr0n: that is why we don't like GUIs, they hide too much information from you. |
| 16:24.09 | nybbled | lirakis_work, ManxPower: thanks for the help guys. I'll give this a try now. Should i enable canreinvite on all the sip peers aswell? |
| 16:24.17 | ManxPower | cr0n: what happens if you try to call a number on that PRI? |
| 16:24.30 | ManxPower | (from a mobile phone) |
| 16:24.31 | lirakis_work | nybbled: yes |
| 16:24.40 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.142) |
| 16:24.58 | cr0n | ManxPower: call comes in no problem, i can dial any one of the numbers specified to it from a mobile or a landline, all good |
| 16:25.11 | *** join/#asterisk bminish (n=bminish@89.19.69.28) |
| 16:25.34 | ManxPower | cr0n: put the contents of /etc/asterisk/zapata.conf on pastebin.ca |
| 16:25.45 | cr0n | ManxPower: i agree about guis, im busy doing a clean install on a second box as well, but annoys me that it will not work here either |
| 16:26.23 | cr0n | ManxPower: http://www.pastebin.ca/977053 |
| 16:27.23 | [TK]D-Fender | cr0n: jsut a thought, try "pridialplan=national" and I think there was another called prilocaldialplan |
| 16:27.37 | [TK]D-Fender | cr0n: After following whatever ManxPower has advised first |
| 16:27.41 | *** join/#asterisk qdk_ (n=qdk@195.242.194.42) |
| 16:28.45 | aiurea | could someone help me with this http://pastebin.com/m6bee23e2 ? I receive invalid SDP information, however the SDP appears to be OK |
| 16:28.54 | ManxPower | cr0n: I cannot help you unless you un-GUI your config files and stop using #include |
| 16:29.09 | ManxPower | But I am out of time. |
| 16:29.20 | cr0n | ManxPower: agreed |
| 16:30.30 | cr0n | [TK]D-Fender: giving that a shot |
| 16:31.22 | nybbled | lirakis_work: ok cool. thanks! |
| 16:33.56 | lirakis_work | oy .. 7 terminal sessions open .. + screen .. way to much stuff processing |
| 16:35.12 | Jumpie | lol ManxPower |
| 16:35.35 | aiurea | nobody? |
| 16:35.38 | cr0n | [TK]D-Fender: tried a few things, still nothing.. including: 'pridialplan=national' and then 'prilocaldialplan=unknown' etc. |
| 16:37.09 | *** join/#asterisk Greek-Boy (n=email@41.221.58.13) |
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| 16:39.08 | *** part/#asterisk jivco (n=jivco@85.187.217.6) |
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| 16:43.28 | [TK]D-Fender | cr0n: Ok, take a look at "pri intense debug" at CLI and talk with your telco ans see if you can spot anything. |
| 16:43.58 | cr0n | [TK]D-Fender: thank you |
| 16:44.30 | *** part/#asterisk matozzao (n=matozzao@201.76.228.176) |
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| 16:54.15 | aiurea | I got disconnected. Can anyone help me with the Invalid SDP message? |
| 16:54.24 | ManxPower | I suspect that the #included files are overwriting the main config file |
| 16:54.28 | UnixDog | having dsl issues |
| 16:54.34 | UnixDog | rickross: back |
| 16:55.17 | ManxPower | aiurea: Results 1 - 10 of about 533 from lists.digium.com for Invalid SDP AND "too lazy to use google" |
| 16:55.21 | kamanashisroy | hi, I understand that a registered user can log in a queue, But I am trying to load queue members realtime .. I just cannot get how a queue member is called .. any hing ? |
| 16:56.06 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 16:56.09 | *** join/#asterisk joe (n=nnnnnnjs@ip66-107-33-195.z33-107-66.customer.algx.net) |
| 16:56.32 | aiurea | ManxPower: i appreciate the "help". Of course I looked there before. |
| 16:57.17 | ManxPower | cr0n: here is my entire /etc/asterisk/zapata.conf file: http://pastebin.ca/977091 |
| 16:57.30 | dandre | Hello, |
| 16:57.54 | dandre | Is there any server side phone book for asterisk? |
| 16:58.07 | [TK]D-Fender | dandre: Feel free to go make one. |
| 16:58.13 | ManxPower | cr0n: I would turn on echocancel and echotraining, as we use an external EC box and so don't need Asterisk doing the EC |
| 16:58.49 | ManxPower | dandre: It would be pretty trivial to make a server side address book. The hard part would be getting IP Phone vendors to support it. |
| 16:58.53 | dandre | sure but if there is one available it could help |
| 16:59.15 | ManxPower | dandre: address books are considered "provisioning" and should be done in the phone. |
| 17:00.21 | dandre | there are possibilities to access an xml remote adress book from a phone |
| 17:00.45 | [TK]D-Fender | dandre: Clearly depends on the phone. Has nothing to do with * |
| 17:00.48 | ManxPower | dandre: from WHICH brand of phone, they all do it differently -- there is no real standard |
| 17:01.06 | ManxPower | ~phones |
| 17:01.07 | jbot | phones is, like, http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. places like such as |
| 17:01.31 | dandre | ok I didnt kwon, I use st2030 and 2022 from thomson |
| 17:01.53 | ManxPower | dandre: you would have to check with Thomson to find out. |
| 17:03.16 | ManxPower | Polycom supports XML directories. I don't know about the others, but I bet Aastra does. |
| 17:04.03 | [TK]D-Fender | ManxPower: Though what he's looking for is centralized, so that'd be more a MicroBrowser thing. |
| 17:05.12 | *** join/#asterisk JayTee52 (n=jforde@207-67-84-177.static.twtelecom.net) |
| 17:06.38 | JayTee52 | wow, I just discovered that Polycom phones show up in my Cisco Network Assistant. |
| 17:07.17 | *** join/#asterisk bmg505 (n=leon@196-209-79-247-tbnb-esr-2.dynamic.isadsl.co.za) |
| 17:10.01 | *** join/#asterisk Parantido (n=Parantid@host-84-221-10-66.cust-adsl.tiscali.it) |
| 17:10.12 | Parantido | hi |
| 17:16.27 | *** part/#asterisk shtoom (n=shtoom@121.246.167.147) |
| 17:17.29 | *** join/#asterisk asdx (n=diego@adsl-157-61.click.com.py) |
| 17:17.33 | asdx | hi |
| 17:17.49 | asdx | what should i do if i want to use SIP and the telco/internet provider is blocking it? |
| 17:19.03 | UnixDog | they are blocking 5060 ? |
| 17:19.07 | UnixDog | are you shure |
| 17:19.16 | UnixDog | are you shure its not a nat issue |
| 17:19.26 | UnixDog | you can use a sip proxy |
| 17:19.42 | asdx | a lot of people been saying that they block SIP |
| 17:20.22 | UnixDog | if they block then find a good sip proxy |
| 17:20.24 | asdx | like openser? |
| 17:20.31 | UnixDog | yeah |
| 17:20.40 | Qwell | and how are you going to get to the proxy? |
| 17:20.44 | Qwell | if...sip is...blocked |
| 17:20.46 | asdx | how can i know if is a nat issue or port blocking? |
| 17:20.55 | UnixDog | and see if you voip provider offers connections on a diff port |
| 17:22.24 | asdx | ok |
| 17:23.04 | *** join/#asterisk adrin_ (n=adrin@chello089078221161.chello.pl) |
| 17:24.19 | UnixDog | try using iax if your provider offers it |
| 17:24.32 | asdx | i use iax |
| 17:25.16 | asdx | but i believe that the users i'm configuring asterisk for, are going to have sip devices, etc |
| 17:25.26 | asdx | and i need a way to test it from here |
| 17:25.29 | asdx | with sip |
| 17:25.35 | adrin_ | hmm i have a question concerning SIP and NAT and asterisk: i have an asterisk server standing behind the nat and a sip peer that is behind internet and another nats, i have control over both routers so i can redirect ports if neccessary... is there a way to get calls working (i can register but i have no sound), what is the proper setup, should i set nat=yes? in what sections? general? [peer]? the route is peer<->NAT1<->Internet<->NAT2<->as |
| 17:25.35 | adrin_ | terisk |
| 17:26.06 | UnixDog | in yous sip.cfg put |
| 17:26.07 | Shotygun | adrin_: Port map the SIP port, 5060, and port map the rtp ports |
| 17:26.17 | UnixDog | extenip or extenhost = |
| 17:26.28 | UnixDog | localnet=ip/netmask |
| 17:26.33 | UnixDog | nat=yes |
| 17:26.57 | Shotygun | and as said, use externip |
| 17:27.23 | UnixDog | externhost if you use dyndns |
| 17:27.34 | UnixDog | that way it adjust for the ip changes |
| 17:27.38 | [TK]D-Fender | adrin_: Read up : |
| 17:27.40 | [TK]D-Fender | ~sipnat |
| 17:27.41 | 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 |
| 17:27.42 | [TK]D-Fender | ^^^^^^^^ |
| 17:30.05 | adrin_ | thanks |
| 17:30.08 | adrin_ | ok |
| 17:30.15 | adrin_ | but i have to redirect rtp ports too? |
| 17:31.01 | ManxPower | adrin_: only if you want to hear audio 8-) |
| 17:31.13 | adrin_ | lol |
| 17:31.35 | ManxPower | adrin_: SIP, MGCP, SCCP/Skinny, and H323 ALL use RTP to transport audio. The only thing different is the SIGNALING (call setup/teardown/etc) |
| 17:31.51 | adrin_ | hmm ok |
| 17:32.19 | adrin_ | ok but ihave to open ports only on server? |
| 17:32.48 | ManxPower | correct |
| 17:33.38 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
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| 17:35.34 | adrin_ | ok thanks |
| 17:35.52 | adrin_ | oooh one more question ... how do i set these ports to a certain range? |
| 17:35.57 | adrin_ | is it something predictable |
| 17:36.06 | adrin_ | or do i have to set it explicitly in sip.conf |
| 17:36.55 | ManxPower | adrin_: did you NOT read the Quick guide on configuring * + SIP behind NAT |
| 17:36.56 | *** join/#asterisk shinao1 (n=shinao1@41.221.165.43) |
| 17:37.24 | adrin_ | ok sorry |
| 17:37.36 | ManxPower | It's pretty obvious you did not. |
| 17:37.40 | ManxPower | go read |
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| 17:42.05 | Greek-Boy | i dont get it. the extension/context that i am trying to call through a IAX2 trunk exists yet the call gets rejected with CAUSE CODE 3 :( |
| 17:42.44 | *** join/#asterisk hohum (i=dcorbe@apollo.wavelen.net) |
| 17:43.30 | ManxPower | Greek-Boy: that is No Route to Destination. You have a networking or address issue |
| 17:44.03 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 17:44.03 | Corydon76-dig | ManxPower: No, the context/extension doesn't exist |
| 17:44.18 | tristanbob_ | anyone know the difference between the Digium TE120P, TE121, TE122? |
| 17:44.23 | Corydon76-dig | Greek-Boy: try: dialplan show extension@context |
| 17:44.23 | Greek-Boy | ManxPower: iax2 show peers shows that everything is good on the network side |
| 17:44.57 | Corydon76-dig | tristanbob_: 122 replaced the 120. 121 is a PCI Express, 122 is PCI |
| 17:45.01 | Greek-Boy | Corydon76-dig: it shows it |
| 17:45.06 | *** join/#asterisk tobias (n=tobias@cpe-066-026-085-055.nc.res.rr.com) |
| 17:45.08 | ManxPower | Greek-Boy: then chances are you typed something wrong |
| 17:45.12 | Corydon76-dig | Greek-Boy: does it show a priority 1? |
| 17:45.21 | Greek-Boy | Corydon76-dig: yes |
| 17:45.34 | Corydon76-dig | Then I agree with Manx, you typed something wrong |
| 17:45.41 | Corydon76-dig | Greek-Boy: check for extra spaces |
| 17:45.59 | Corydon76-dig | Missing or transposed letters, too |
| 17:46.08 | Greek-Boy | I even put the extension in a seperate context, retyped it without a macro |
| 17:46.19 | tristanbob_ | Corydon76-dig, TE120P is still on the digium website - is it end-of-sale? |
| 17:46.28 | Corydon76-dig | Greek-Boy: are you asking for a permitted context? |
| 17:46.38 | Greek-Boy | yes |
| 17:46.40 | tristanbob_ | http://www.digium.com/en/products/digital/te120p.php |
| 17:46.52 | Greek-Boy | I set it in my iax2.conf |
| 17:46.55 | ManxPower | Greek-Boy: paste JUST the Dial line to start with. |
| 17:46.57 | Greek-Boy | iax.conf even |
| 17:47.19 | ManxPower | Actually paste both the Dial command as shown in the CLI and that same Dial line from extensions.conf |
| 17:48.35 | Kyoshi | using freetds, i can get cdr's written to mssql just fine, but when trying to config realtime config from the same sqldb, asterisk fails and reverts back to the .conf files. how can i see why it's failing? |
| 17:49.02 | Corydon76-dig | Greek-Boy: are you using 1.6? |
| 17:49.33 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 17:51.30 | ManxPower | Greek-Boy: unfortunately I do not have all day to work with you on this. You might want to respond to requests, information, and suggestions faster. |
| 17:54.54 | *** join/#asterisk pokerfacepenguin (n=joe@68.16.15.79) |
| 17:55.02 | Greek-Boy | ManxPower: http://pastebin.com/d5568d8b |
| 17:55.34 | Greek-Boy | ManxPower: My apologies, was cutting pasting. |
| 17:55.52 | Greek-Boy | ManxPower: will practice my cutting/pasting skills |
| 17:58.48 | Greek-Boy | ManxPower: BoxB is at a telco provider that has a PRI connected to it. When a call comes from the PSTN BoxB strips 7 digits off and passes it to BoxA. |
| 17:59.49 | Greek-Boy | so when I call 0797120712 BoxB dials 712 to BoxA. i have made sure that the context contains that extension. |
| 18:02.24 | [TK]D-Fender | Greek-Boy: And you made sure not to SHOW us these contexts. |
| 18:03.36 | Greek-Boy | [TK]D-Fender the dial lines are in my paste |
| 18:04.02 | [TK]D-Fender | Greek-Boy: I don't care about the dial lines right now. Show me that the COMPLETE picture is perfect. |
| 18:04.38 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:04.42 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 18:05.01 | *** join/#asterisk nightsapper (n=wcollier@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 18:05.45 | *** join/#asterisk wonderworld (n=voici@ip-62-143-163-248.1211G-CUD12K-01.ish.de) |
| 18:06.27 | *** join/#asterisk MasterBob (n=Weebol@unaffiliated/masterbob) |
| 18:07.32 | UnixDog | rickross: you there |
| 18:07.44 | rickross | hi UnixDog, yep |
| 18:07.48 | UnixDog | <digitmap dialplan.digitmap="*.T|#.T|xxxx|[2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx" dialplan.digitmap.timeOut="3|3|3|3|3|3|3|3"/> |
| 18:07.56 | *** join/#asterisk riddlebox (n=riddlebo@64.123.54.41) |
| 18:08.02 | UnixDog | that will fix the dial plan |
| 18:08.09 | UnixDog | in 3.0.0.0 |
| 18:08.16 | UnixDog | to many 0 |
| 18:08.33 | UnixDog | just add xxx| if you need 3 digit |
| 18:08.36 | rickross | thx, we're quite close to that now - we have 3-digit extensions |
| 18:08.42 | rickross | exactly |
| 18:08.48 | Katty | sits and squeaks a doggy toy. |
| 18:08.52 | rickross | what is the "#.T" for? |
| 18:09.03 | Katty | funny thing about picking up pet toys at the store... is that you always want to squeak them. |
| 18:09.19 | UnixDog | to dial #and any digits |
| 18:09.23 | UnixDog | T = timeout |
| 18:09.36 | Greek-Boy | Corydon76-dig: sorry just saw your msg now. No i'm not using 1.6. I'm on 1.4.18.1 planning to uprade to 1.4.19 soon |
| 18:09.53 | Katty | squeaks toy in [TK]D-Fender's general direction. |
| 18:10.08 | rickross | wouldn't it need to be "#xx.T" |
| 18:10.20 | rickross | to get at least 1 char after the # ? |
| 18:10.39 | UnixDog | nope |
| 18:10.58 | ManxPower | Greek-Boy: why are you asking for help and then leaving? |
| 18:11.05 | ManxPower | If you want help, then take the time to get help |
| 18:11.23 | Katty | hi manx! |
| 18:11.28 | Katty | hugs ManxPower |
| 18:11.48 | UnixDog | but #.T wont work unless you use #infront of extensions |
| 18:11.54 | ManxPower | Hello, Katty |
| 18:12.07 | UnixDog | and I dint know many pbx systems that do |
| 18:12.10 | ManxPower | UnixDog: you want as few Ts as possible |
| 18:12.13 | UnixDog | I use it for speed dials |
| 18:12.19 | rickross | isn't the "." for zero or more of the preceding char? |
| 18:12.19 | Greek-Boy | ManxPower: I'me here |
| 18:12.46 | ManxPower | rickross: you could read the MGCP RFC, which is what Polycom based it's dialplan config on. |
| 18:13.02 | rickross | ManxPower - thanks a million ;) |
| 18:13.03 | ManxPower | Greek-Boy: so give us the information we asked for 20 mins ago |
| 18:13.10 | *** join/#asterisk mking_cd (n=mking_cd@pool-72-78-131-132.phlapa.east.verizon.net) |
| 18:13.19 | rickross | I did, however, read the polycom docs and quoted it here an hour ago |
| 18:13.21 | ManxPower | rickross: ignore all of the RFC, except the dialplan stuff |
| 18:13.40 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 18:13.56 | mking_cd | I'm looking for a way to log a queue MEMBER name in the filename of a queue call recorded with MixMonitor. Can anyone help? |
| 18:14.03 | ManxPower | rickross: must be new docs. The Polycom docs I have basically say "go read the MGCP RFC" in the section covering dialplans |
| 18:14.07 | rickross | hopefully their docs should be sufficient to avoid the need to read the RFC, too! |
| 18:14.11 | Greek-Boy | ManxPower: did u see the paste? I will paste the dial plan too |
| 18:14.28 | rickross | ManxPower - the docs were spare, for sure |
| 18:14.37 | rickross | but they specifically speak to the "." char |
| 18:14.46 | *** part/#asterisk pokerfacepenguin (n=joe@68.16.15.79) |
| 18:15.18 | ManxPower | Greek-Boy: "Dial("Zap/21-1", "IAX2/oi-asterisk01/712")" I need to see the iax.conf entry for [oi-asterisk01] in iax.conf on both servers |
| 18:17.09 | rickross | A period (".") matches an arbitrary number, including zero, of occurrences of the preceding construct |
| 18:18.25 | [hC] | rickross: heh, i looked this up LAST night. Silly me presumed that a . just meant "any number of occurrences of any character" |
| 18:19.07 | [hC] | rickross: but there is the very distinct mention, there, that its any number of occurrences of the the character or expression just before the "." |
| 18:19.19 | Greek-Boy | ManxPower: http://pastebin.com/d369120fd shows dial plans |
| 18:19.20 | rickross | it may - the Polycom docs might be incorrect (if the RFC specifies some other meaning) |
| 18:19.33 | ManxPower | rickross: where in ASTERISK . means match "one or more" |
| 18:19.38 | tristanbob_ | Does anyone know what T1 card the Switchvox AA60 supports? |
| 18:19.50 | Greek-Boy | ManxPower: in the first paste on http://pastebin.com/d5568d8b i have pasted iax.conf as well for both boxes. |
| 18:20.00 | ManxPower | tristanbob_: why don't you ask on the switchvox channel? |
| 18:20.52 | *** join/#asterisk jjshoe (n=jjshoe@72.37.252.50) |
| 18:21.21 | ManxPower | Greek-Boy: remove rsa auth stuff when troubleshooting. Also your context= lines don't match between extensions.conf and iax.conf. |
| 18:22.39 | Greek-Boy | ManxPower: I changed to context bolbol |
| 18:22.50 | Greek-Boy | ManxPower: so that I could test without the macro |
| 18:23.13 | Greek-Boy | I updated iax.conf on BoxB to use bolbol |
| 18:23.20 | *** join/#asterisk asdx (n=diego@adsl-157-61.click.com.py) |
| 18:23.25 | ManxPower | Greek-Boy: whatever context= is set to in iax.conf is the context in extensions.conf that the call will land in. |
| 18:23.40 | asdx | ok, i have to setup an asterisk server for some guy, what would be better to use, SIP or IAX2? |
| 18:23.48 | mking_cd | are there any channel variables that would provide me the member name in the queue() channel? |
| 18:23.48 | asdx | he told me he will just use softphones for now |
| 18:23.57 | Greek-Boy | I mean I updated iax.conf on BoxA to use bolbol for friend BoxB |
| 18:23.57 | asdx | so ig uess IAX will be easier for now, right? |
| 18:24.02 | ManxPower | so if you have a context=billybob in sip.conf then you must have a [billybob] section in extensions.conf |
| 18:24.25 | ManxPower | Greek-Boy: it sounds to me that the info in your pastebin.ca is not accurate. |
| 18:24.29 | Greek-Boy | ManxPower: I have verified that the contexts do exist and do match up yet I am still getting that error |
| 18:24.39 | Greek-Boy | ManxPower: I will remove rsa and try again |
| 18:25.15 | ManxPower | Greek-Boy: also remove trunk=yes |
| 18:25.19 | ManxPower | set a secret too |
| 18:25.24 | UnixDog | back |
| 18:25.53 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
| 18:26.06 | asdx | if my users are going to use softphones for now, IAX2 would be enough righ? |
| 18:26.09 | asdx | right* |
| 18:26.34 | Greek-Boy | ManxPower: got it |
| 18:26.52 | tristanbob_ | ManxPower, already did - that channel is dead! |
| 18:26.53 | UnixDog | rickross: the t makes it time out faster so when you dial 4 digits or 3 digits fallowed by a t it dials right away |
| 18:27.09 | rickross | UnixDog: thx |
| 18:27.11 | asdx | i'm not phisically there so i don't want to deal with NAT issues |
| 18:27.14 | ManxPower | tristanbob_: That is the correct place. If you can't get support for Switchvox then why are you considering using it? |
| 18:27.27 | ManxPower | asdx: IAX2 has NAT issues too. |
| 18:27.57 | ManxPower | tristanbob_: I suspect not a single person on this channel uses Switchvox |
| 18:28.07 | asdx | ManxPower: so what do you suggest |
| 18:28.19 | ManxPower | asdx: I suggest you do the work to make SIP work. |
| 18:28.29 | asdx | ManxPower: ok |
| 18:28.48 | tristanbob_ | ManxPower, switchvox uses asterisk, so we all have that in common |
| 18:29.40 | ManxPower | tristanbob_: And Windows 3.11 is like DOS -- it uses DOS afterall. But you would not go to a Windows channel for support for DOS, would you? |
| 18:29.42 | ManxPower | ~freepbx |
| 18:29.43 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 18:29.43 | ManxPower | ~amp |
| 18:29.44 | jbot | i heard amp is NOT supported here! People using it should join #freepbx (FreePBX is the new name of AMP) |
| 18:29.52 | ManxPower | ~switchvox |
| 18:30.21 | *** part/#asterisk MasterBob (n=Weebol@unaffiliated/masterbob) |
| 18:30.33 | ManxPower | tristanbob_: GUIs for Asterisk have incredibly complicated config files -- not something worth supporting here, as this channel is for ASTERISK, not Asterisk + GUI |
| 18:30.37 | jameswf-home | not long until find hot women for sex in your area http://files.jameswf.info/tbnsfw.png |
| 18:30.39 | jjshoe | I used switchvox for a day, it was cute when they refused to give up the root password to the box. |
| 18:31.06 | jameswf-home | jjshoe: just change it |
| 18:31.15 | asdx | ManxPower: the problem is also that my telco blocks SIP |
| 18:31.24 | ManxPower | asdx: then you must use IAX |
| 18:31.32 | jjshoe | jameswf-home obviously I booted to single user mode, modified pam, and did such. |
| 18:31.44 | *** join/#asterisk rupa (i=rupa@gw.rupa.com) |
| 18:32.01 | asdx | ManxPower: i want to use SIP as is more widely supported, and in that aspect it will make everyone life's easier |
| 18:32.06 | asdx | ManxPower: i'll see what i can do |
| 18:32.16 | ManxPower | asdx: if your telco blocks SIP it won't be easier |
| 18:32.27 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
| 18:32.51 | asdx | ManxPower: but the voip users are somewhere else in the world |
| 18:32.52 | ManxPower | If your telco is smart it will do deep packet inspection and that would mean no matter what port you changed it to, it would be blocked. I don't know of many ISPs that do that. |
| 18:32.59 | asdx | ManxPower: they wont go through my telco |
| 18:33.04 | asdx | ManxPower: just me |
| 18:33.18 | ManxPower | asdx: then use SIP. |
| 18:33.29 | *** join/#asterisk Skarmeth (n=Skarmeth@iris.aspec.com.br) |
| 18:33.33 | ManxPower | I don't CARE. You know what your options are -- pick one. |
| 18:34.09 | asdx | can i use IAX for my part, and SIP for them, and still talk with them? |
| 18:34.37 | ManxPower | you can use any mix of protocols |
| 18:34.42 | asdx | nice |
| 18:34.51 | asdx | thats what i'm going to do then |
| 18:38.34 | Skarmeth | hi all |
| 18:39.35 | Skarmeth | anyone having troubles unloading wct4xxp module? it seems to me that it don't update usage counters... |
| 18:40.32 | rupa | anyone using voicepulse of incoming calls receiving caller id *name* information? I spoke to support and they claim to do a database dip for the name, but the rpid that I see in the sip debug shows the # not the name. Just wondering if anyone has seen a callerid *name* from voicepulse. |
| 18:41.13 | rupa | alternatively, any did provider that you KNOW does the database dip? |
| 18:43.27 | *** join/#asterisk colinm_ (n=colinm@VDSL-130-13-122-158.PHNX.QWEST.NET) |
| 18:43.38 | *** join/#asterisk Greek-Boy (n=email@41.221.58.13) |
| 18:46.09 | Skarmeth | When I try to remove the kermel module, I get a kernel oops (http://pastebin.com/m424fdc2c). I am using Fedora 8 x64 with all updates on a Dell Optiplex 745 Core 2 Duo (this is a development system). |
| 18:49.13 | *** join/#asterisk dacs (n=haiger@unaffiliated/dacs) |
| 18:52.06 | ManxPower | don't do that 8-) |
| 18:54.16 | *** join/#asterisk grandpapadot (n=null@mail.heavylogic.com) |
| 18:54.29 | grandpapadot | Hi all. Any general recommendations for a good SIP prioritizing firewalls? |
| 18:55.22 | UnixDog | milkfish |
| 18:55.39 | UnixDog | dd-wrt |
| 18:55.45 | grandpapadot | Commercial/Appliance, rather. |
| 18:59.15 | *** join/#asterisk ZaVoid (n=zavoid@66-95-182-90.client.dsl.net) |
| 19:00.43 | *** join/#asterisk C4colo (n=DJpyro@67.41.154.214) |
| 19:01.00 | C4colo | is there a way to point the zaptel make process to the kernel sources? |
| 19:01.03 | C4colo | well, headers anyway |
| 19:01.26 | C4colo | yum claims that the kernel-devel packages are installed for the kernel, but zaptel errors out and says that they are not |
| 19:02.44 | C4colo | is there a --kernel-src-dir switch or something? |
| 19:04.57 | asdx | how can i see if there is some call going on? |
| 19:06.35 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 19:07.10 | DarKnesS_WolF | PSTN line has CID --->> PBX ----> Asterisk ----> SIP phone don't get the CallerID how can i pass the caller ID ? |
| 19:07.41 | rupa | DarKnesS_WolF, does asterisk see the callerid? |
| 19:07.56 | asdx | http://pastebin.com/m418fddba |
| 19:07.58 | asdx | i'm getting that |
| 19:07.59 | asdx | why? |
| 19:08.40 | DarKnesS_WolF | rupa: no |
| 19:08.49 | rupa | how is the pbx interfaced to asterisk? |
| 19:09.00 | DarKnesS_WolF | rupa: FXS ----> FXO in asterisk |
| 19:09.28 | Claude2008 | Any one available to help (I don't mind paying someone just need to get this resolved) |
| 19:09.32 | rupa | you use the pbx is passing callerid as FSK tones between ring 1 and 2? |
| 19:09.46 | rupa | Claude2008, just ask, if someone can help you might even get it for free |
| 19:10.59 | DarKnesS_WolF | rupa: don't know :-D i didn't configure the PBX :-D conf. rupa the asterisk works fine with CID if i pluged normal PSTN directly |
| 19:11.39 | rupa | so I would suspect the PBX isn't passing the caller id properly. can you try a regular analog CID enabled phone on that line to see if callerid is passed from the PBX? |
| 19:11.46 | Claude2008 | i just formated my box, and installed the latest Trixbox 2.6.0.7 |
| 19:12.06 | Claude2008 | and i am using Sangoma A200 with 6 fxo |
| 19:12.12 | DarKnesS_WolF | rupa: ok i'll test that the pbx not infront of me right now |
| 19:12.17 | rupa | ~trixbox |
| 19:12.18 | jbot | [~trixbox] trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org |
| 19:12.24 | ManxPower | Claude2008: you need to be in #trixbox, not #asterisk |
| 19:12.42 | DarKnesS_WolF | rupa: but i think the operater phone gets the CID normaly |
| 19:12.47 | Claude2008 | since i formated, i cannot get zap channles to work |
| 19:13.10 | Claude2008 | currently no one is in trixbox channle |
| 19:13.15 | Claude2008 | i guess i have to wait |
| 19:13.21 | UnixDog | ManxPower: you need to add PBXinAFLash to the list of distros |
| 19:13.26 | asdx | hi |
| 19:13.30 | rupa | DarKnesS_WolF, is the oper phone an analog phone or a digital pbx phone? I imagine the pbx is getting cid from the pstn, but isn't passing cid out the FXS. |
| 19:13.32 | asdx | my boss can't hear anyone on his calls |
| 19:13.35 | asdx | [Apr 8 14:10:59] NOTICE[30686]: chan_sip.c:5481 process_sdp: No compatible codecs, not accepting this offer! |
| 19:13.39 | asdx | i'm getting that on the server |
| 19:13.43 | asdx | what can i do? |
| 19:13.47 | UnixDog | you need to set codecs on the trunks |
| 19:13.52 | UnixDog | and on the extensions |
| 19:13.56 | UnixDog | that you allow |
| 19:14.01 | asdx | i have |
| 19:14.03 | asdx | disallow=all |
| 19:14.05 | asdx | allow=gsm |
| 19:14.07 | UnixDog | the answer is right on the error line |
| 19:14.17 | UnixDog | does the phone support gsm |
| 19:14.23 | asdx | zoiper |
| 19:14.25 | UnixDog | does the provider support gsm |
| 19:14.25 | asdx | i think so |
| 19:14.28 | *** join/#asterisk NoamRotter (n=noam@mail.browarnik.com) |
| 19:14.28 | asdx | i was using fine yesterday |
| 19:14.31 | rupa | anyone using voicepulse of incoming calls receiving caller id *name* information? I spoke to support and they claim to do a database dip for the name, but the rpid that I see in the sip debug shows the # not the name. Just wondering if anyone has seen a callerid *name* from voicepulse. |
| 19:14.32 | DarKnesS_WolF | rupa: digital one |
| 19:14.36 | asdx | UnixDog: it's a user to user thing |
| 19:14.40 | asdx | UnixDog: no voip provider involved |
| 19:14.41 | asdx | yet |
| 19:14.58 | DarKnesS_WolF | asdx: try with ulaw |
| 19:15.13 | asdx | very simlpe config |
| 19:15.18 | jblack | 9164401664 bytes (9.2 GB) copied, 0 seconds, Infinity B/s |
| 19:15.27 | UnixDog | are you on the same net work or is this over nat |
| 19:15.30 | rupa | DarKnesS_WolF, right, so the PBX may not support CID on analog FXS lines. Or maybe there is an optino to set, but I doubt the problem is asterisk. Try the analog phone on the FXS. Technically it doesn't have to be a CID phone. Answer on first right and listen for the FSK tones between ring 1 and 2. |
| 19:15.33 | asdx | http://pastebin.com/m73fe7352 |
| 19:15.35 | asdx | that's my config |
| 19:15.45 | UnixDog | is this over nat ? |
| 19:15.51 | DarKnesS_WolF | rupa: thx will do |
| 19:16.01 | asdx | UnixDog: yes, i think so |
| 19:16.04 | *** join/#asterisk s0lid (n=s0lid@125.252.90.128) |
| 19:16.07 | asdx | UnixDog: they might have a router there |
| 19:16.10 | UnixDog | ok |
| 19:16.14 | UnixDog | in sip.conf |
| 19:16.19 | UnixDog | you need a few lines |
| 19:16.24 | UnixDog | ~nat |
| 19:16.24 | jbot | [nat] Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 19:16.34 | asdx | UnixDog: nat=yes and stuff? |
| 19:16.51 | UnixDog | localnet=ip.of.network/netmask |
| 19:17.02 | UnixDog | extenhost= or externip= |
| 19:17.13 | [TK]D-Fender | asdx: 1st guess, ~sipnat |
| 19:17.16 | UnixDog | depending on if its a static or dynamic setup |
| 19:17.21 | [TK]D-Fender | ~sipnat |
| 19:17.22 | 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 |
| 19:17.23 | [TK]D-Fender | ^^^^^^^^^^^^^ |
| 19:17.31 | asdx | damn this guy is being inpatient on me, and i'm a noob |
| 19:17.37 | *** join/#asterisk nybbled (n=nybbled@about/apple/performa/nybble) |
| 19:17.40 | nightsapper | CNAM not delivered by any VoIP provder yet? |
| 19:17.44 | nybbled | hey all, anyone here have experience with using canreinvite=update ? |
| 19:17.45 | UnixDog | no such thing as a noob |
| 19:17.49 | UnixDog | not anymore |
| 19:17.53 | asdx | ok |
| 19:17.57 | UnixDog | that term has been worn out |
| 19:18.02 | nybbled | nightsapper: CNAM would be nice |
| 19:18.10 | nightsapper | Only noob use the term noob. ACK! |
| 19:18.48 | nightsapper | Okies. THanks - I JFGI but no real answeres outthere for CNAM on a VoIP |
| 19:19.06 | rupa | nightsapper, voicepulse claims they did when I signed up, but I have no evidence of it being done |
| 19:19.53 | nightsapper | ANI/DNIS delivered withthe call, CNAM supposed to come between ring 1 and 2 in general. |
| 19:20.01 | asdx | so i add nat=yes and ullaw? |
| 19:20.05 | asdx | and also the rest of the options for nat |
| 19:20.47 | nightsapper | or at least thats how it works on POTS. Not so sure on PRI (be coming over the D channel, but its def the provider that has to do the lookup and fill+send the CNAM) |
| 19:21.13 | rupa | nightsapper, right, the provider has to do the database dip. I would think they'd do it prior to setting up the call to me |
| 19:21.20 | nybbled | nightsapper: its hte provider who does the lookup for you |
| 19:21.43 | rupa | then then using rpid to present the name, the number is in the From: header |
| 19:22.34 | rupa | no name is going to kill this project on wife acceptance testing |
| 19:22.44 | asdx | <PROTECTED> |
| 19:22.48 | asdx | that's his codec |
| 19:22.50 | asdx | ilbc20 |
| 19:23.09 | UnixDog | asdx: http://pastebin.com/m44591db7 |
| 19:23.14 | asdx | <PROTECTED> |
| 19:23.15 | nightsapper | rupa: true dat. Also a big issue for a law office I jsut installed a 53 seat T-1/PRI Asterisk in. |
| 19:23.41 | asdx | UnixDog: thanks |
| 19:23.50 | nightsapper | That and intercom using the phone speakers, and they also wanted a gui to manage the call park. |
| 19:23.53 | [TK]D-Fender | asdx: read the GUIDE. |
| 19:24.01 | rupa | the problem is that there is a cost per database dip, but I'd be willing to pay that on a pass-through basis. |
| 19:24.03 | nightsapper | and a pony too |
| 19:24.07 | asdx | [TK]D-Fender: i will, i will |
| 19:24.44 | rupa | intercom on phone speakers, I think you can do that with something like a slimserver and some scripting... |
| 19:24.53 | nightsapper | rupa -yep. I guess the reason for hte dealeyd DB dip is to prevent call progress form getting hung on a DB issue. ANI/DNIS are baseline protocol. CNAM isnt. |
| 19:25.12 | rupa | true |
| 19:25.37 | rupa | or put a sound card in your * server and play through that |
| 19:26.21 | asdx | can i say, allow=foo,bar |
| 19:26.27 | nightsapper | 57 Aastri 53i in this place. having to fake SLA with a bridge (everyone has a private DID, and the Attornys have a "public" did that rings at the secretary phone and teir phone at the same time). |
| 19:26.28 | asdx | or i have to allow=foo allow=bar ? |
| 19:27.34 | asdx | allow=foo allow=bar |
| 19:27.34 | UnixDog | asdx go get asterisk-now |
| 19:27.34 | UnixDog | and use it |
| 19:27.34 | asdx | no |
| 19:27.34 | asdx | i don't like gui |
| 19:27.34 | UnixDog | it will do all the hard work for you |
| 19:27.45 | asdx | i want to learn |
| 19:28.18 | asdx | if i wanted to use a gui i would have used that in the first place, but i don't want to |
| 19:28.52 | nightsapper | lucky they had a sane plan on the old mini-PBX. Nxx-1100 thru Nxx-1199 1-to-1 to ext 100-199, plenty of spares. |
| 19:29.03 | *** join/#asterisk khaefner (n=khaefner@129.82.174.114) |
| 19:30.46 | khaefner | Greeting everyone! Quick question: In Asterisk 1.6 is SMDI RealTime enabled? There is a mapping from SMDI station to mailbox, is this configurable in a MySQL DB? |
| 19:30.48 | nightsapper | asdx, most use a DB to set up real-world asterisk. And a gui is pretty handy if you want someone else to be able to admin/mantain the system for routine things. |
| 19:32.12 | nightsapper | so don;t be quite so quick to dicard the gui. it will get thigns running then you can freelance off it to learn specific things. And if you get it all horked up, a-now is a good way to get it unhorked. |
| 19:32.20 | asdx | nightsapper: my boss said he wanted to administer asterisk (add users, etc) |
| 19:32.40 | nightsapper | save yourself the time and effort. a-now GUI will simplfy that greatly |
| 19:33.03 | asdx | nightsapper: but the other day i asked that here and someone said "asterisk gui sucks" |
| 19:33.10 | nightsapper | I cna hand code a lot of this stuff, but after a few weeks of it, I was more than happy to use the GUI. :-) |
| 19:33.58 | nightsapper | it may suck, but having to hand process 25-30 MACD requests is a pain too. GUI makes it pretty easy. |
| 19:34.09 | nightsapper | I'm a lazy admin :-) |
| 19:34.18 | asdx | ok, one more question |
| 19:34.20 | asdx | in externip |
| 19:34.26 | nybbled | heh. whats a Gui? LOL |
| 19:34.32 | asdx | should i put the ip address of who? |
| 19:34.44 | asdx | of the server? |
| 19:34.52 | nybbled | asdx: your public ip address that the server would be talking to the internet on |
| 19:34.57 | [TK]D-Fender | asdx: read the guide, its all explained in there |
| 19:35.12 | *** join/#asterisk RoyK (n=roy@ip-230-7-149-91.dialup.ice.no) |
| 19:36.11 | khaefner | Anybody have an answer to the SMDI question or should I ask on the dev channel? |
| 19:36.17 | nightsapper | asdx its all about tunneling RTP thru the NAT - that and gettign SIP the right addresses it needs to set up the RTP streams |
| 19:36.47 | nightsapper | kahef - Im onlyup on 1.4 so Im no help for 1.6 (if you find the answer, be kind and post here too?) |
| 19:37.31 | nightsapper | Lesson 1 for Asterisk: know your protocol. ;-) |
| 19:37.31 | [TK]D-Fender | khaefner: Yes, -dev is a far better place for that |
| 19:37.47 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 19:38.14 | UnixDog | http://pastebin.com/m44591db7http://pastebin.com/m44591db7http://pastebin.com/m44591db7http://pastebin.com/m44591db7http://pastebin.com/m44591db7http://pasbrb |
| 19:38.22 | UnixDog | sorry |
| 19:38.37 | *** join/#asterisk RoyK (n=roy@ip-230-7-149-91.dialup.ice.no) |
| 19:38.56 | asdx | ~sipnat |
| 19:38.56 | 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 |
| 19:40.16 | Skarmeth | posted a bug report about wct4xxp unloading http://bugs.digium.com/view.php?id=12387 |
| 19:43.38 | Jumpie | argh my chair just let out a final hiss of doom and my hydraulics dont work now |
| 19:43.39 | Jumpie | shit |
| 19:43.45 | asdx | nightsapper: uhm, this is confusing |
| 19:44.04 | asdx | nightsapper: my asterisk server is somewhere on the internet |
| 19:44.25 | C4colo | no wonder why they failed Jumpie |
| 19:45.12 | asdx | and these people connects to the server over SIP, they are behind a router/NAT setup |
| 19:45.35 | Jumpie | lol |
| 19:46.57 | [TK]D-Fender | asdx: "somewhere on the internet". Could you be a little MORE VAGUE please? |
| 19:47.27 | [TK]D-Fender | asdx: And for your remote clients behind their own NAT, llok at "B"'s setup |
| 19:47.51 | asdx | 208.86.153.80 |
| 19:47.54 | asdx | thats where the asterisk server is |
| 19:48.06 | [TK]D-Fender | asdx: is that * on a public IP? |
| 19:48.14 | asdx | yes |
| 19:48.14 | nightsapper | YIKES |
| 19:48.24 | asdx | 208.86.153.80 |
| 19:48.27 | asdx | that's the ip |
| 19:48.28 | [TK]D-Fender | asdx: Then you don't need all the localnet stuff at all. |
| 19:48.35 | asdx | ok |
| 19:48.40 | nightsapper | hope you locked it down. Dont need NAT - thats cleint side only in your case |
| 19:48.51 | [TK]D-Fender | asdx: All you need is the peer-specific stuff and to prevent re-invites |
| 19:49.06 | *** join/#asterisk colinm_ (n=colinm@VDSL-130-13-122-158.PHNX.QWEST.NET) |
| 19:49.13 | asdx | so i can remove localnet/externip |
| 19:49.17 | asdx | and leave nat=yes there? |
| 19:49.26 | asdx | in [general] |
| 19:50.14 | nightsapper | not dual homed is it? (one eth to the net, one to the lan on a 10.x.x.x or similar?) |
| 19:50.16 | [TK]D-Fender | asdx: No, all of those gone from [general]. Go do it,and then follow the REST of the guide |
| 19:50.25 | ManxPower | asdx: localnet/externip is for when ASTERISK is behind NAT. |
| 19:50.40 | asdx | ohhhh |
| 19:50.41 | asdx | ok |
| 19:50.51 | asdx | done |
| 19:50.55 | ManxPower | all you need for CLIENTS behind NAT is nat=yes and turn off anything on the remote SIP device that has to do with NAT, EXCEPT "nat keepalive" |
| 19:51.03 | nightsapper | yep - if you go asterisk@192.168.1.1 to ROUTER/NAT to (internet) is when you need that |
| 19:51.17 | ManxPower | If both Asterisk and the phone try to do NAT, then you turn off NAT |
| 19:55.08 | asdx | http://pastebin.com/m61e2a509 |
| 19:55.12 | asdx | thats how my sip.conf looks now |
| 19:55.20 | asdx | it is wrong? |
| 19:56.08 | asdx | goes to rtm |
| 19:56.21 | [TK]D-Fender | asdx: YES |
| 19:57.09 | asdx | :/ |
| 19:58.15 | DarKnesS_WolF | mmmmmm i have something a little bit strange .. i want to sperated 4 users into 2 groups so group 1 can call eachothers and group 2 can call eachohters but group 1 can't call group 2 . and in same time group 1 and 2 are able to talk to group 3 which will have the zap channels .. this should be done by context ? there is no other way ? |
| 19:58.25 | asdx | [TK]D-Fender: whats wrong in my config? |
| 19:59.34 | nightsapper | ditch BOTH the NAT statements. |
| 19:59.47 | nightsapper | not needed |
| 19:59.47 | [TK]D-Fender | asdx: canreinvite=no |
| 19:59.49 | [TK]D-Fender | ; IMPORTANT! phones must not be allowed to attempt to |
| 19:59.50 | [TK]D-Fender | ; directly connect with each other |
| 20:00.02 | coolhp | Does compiling asterisk with DEBUG_CHANNEL_LOCKS and DEBUG_THREADS have any impact on performance ? |
| 20:00.04 | [TK]D-Fender | asdx: What part about this is NOT ^&#@%$ CLEAR? |
| 20:00.20 | nightsapper | true dat. all kinds of mischief if yo dont lock downthe reinvites. |
| 20:00.32 | [TK]D-Fender | wonders if he should make that notice 10x bigger and flashing on that post... |
| 20:01.25 | asdx | nightsapper: isnt the nat=yes needed if you are behind the nat, these two users are behind a nat |
| 20:01.43 | nightsapper | coolhp, probably, especially if you have trheads going, it will add overhead and make contexzt switchign cost you cycles. But if you;re not heavily loaded you probably will nto notice |
| 20:01.56 | nightsapper | Oh sorry, my bad for not readign back |
| 20:02.13 | nightsapper | thought you were still talking about that internet side of your server |
| 20:02.18 | asdx | nightsapper: so, it's fine there right? |
| 20:02.25 | [TK]D-Fender | asdx: under [genera], nat=yes tells Asterisk that IT is behind NAT. Under a phone's entry it tells Asterisk that the PHONE is behind a NAT of its own. |
| 20:02.26 | asdx | nightsapper: no |
| 20:02.42 | nightsapper | blegh, conference call, in a conference room. sales people. yawn. At least they get good snacks. |
| 20:02.43 | *** join/#asterisk ariel_ (n=ariel_@c-66-176-41-202.hsd1.fl.comcast.net) |
| 20:03.05 | nightsapper | beats us admins in the rathole.. slide a pizza uner the door once in a while fo rus. |
| 20:03.08 | asdx | [TK]D-Fender: oh, ok |
| 20:03.14 | [TK]D-Fender | asdx: http://pastebin.com/m566cc701 |
| 20:03.42 | Curus | Someone with a death wish and a love of pain should implement all of ICE for Asterisk |
| 20:03.57 | nightsapper | I'd settle for CNAM |
| 20:04.00 | coolhp | Thanks nightsapper : My main concern is that I'm having random lockups when using addqueumember and removequeuemember |
| 20:04.02 | nightsapper | on a regualr bases. |
| 20:04.08 | nightsapper | ^basis |
| 20:04.15 | ariel_ | hello folks |
| 20:04.26 | coolhp | I've been having them since 1.4.13 but they seem to be getting worse with 1.4.19 |
| 20:04.52 | asdx | [TK]D-Fender: thanks |
| 20:04.58 | asdx | [TK]D-Fender: i will go read the manual now |
| 20:05.16 | [TK]D-Fender | asdx: NOW!?!?!? We've been telling you that for HOURS. |
| 20:05.34 | asdx | yeah but my boss has been asking me a lot of questions and it didn't let me concentrate |
| 20:06.07 | asdx | i read a bit of it already |
| 20:06.16 | nightsapper | recommends telecommuting |
| 20:06.41 | nightsapper | is away: CONferenceCall with Salesweasels |
| 20:06.42 | Nugget | recommends slacking |
| 20:06.55 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 20:07.23 | [TK]D-Fender | asdx: You are wasting our time. Don't ask for our help and then ignore it. |
| 20:07.25 | nightsapper | notes that telecommuting and slacking are not incompatible. (World of Warcraft and troublshooting and Asterisk Server is profitbale and Fun) |
| 20:07.53 | ariel_ | wonders when some developer of asterisk is going to fix the issues with running asterisk -rx "Show Channels" and getting it turncated. |
| 20:07.58 | nightsapper | is away: Reall REALLY away this time |
| 20:09.31 | *** join/#asterisk talntid (n=t@66.208.251.170) |
| 20:09.48 | Curus | the show commands on the console are always a bit risky, you never know what Asterisk might decide to output |
| 20:10.19 | ariel_ | Curus, yes but even the core ones are the same. |
| 20:10.28 | Curus | Yes |
| 20:10.29 | ariel_ | Some times you need info right away |
| 20:10.42 | Curus | I didn't particularly mean that some commands are special, they all have that problem |
| 20:10.58 | asdx | [TK]D-Fender: i don't ignore it |
| 20:11.20 | asdx | sorry |
| 20:11.21 | Curus | The management port can be better |
| 20:11.27 | Curus | Or in some ways worse |
| 20:11.41 | ariel_ | yes but you can't always use the management port |
| 20:12.11 | Curus | No, it would be nice if asterisk had a third interface |
| 20:12.33 | UnixDog | third interfaces ? |
| 20:12.35 | Curus | A command line one, which only returned what it was asked for |
| 20:12.59 | UnixDog | sip show vibrator |
| 20:13.01 | UnixDog | lol |
| 20:13.07 | UnixDog | like that |
| 20:13.12 | ariel_ | lol = it's why I run the command externally |
| 20:13.32 | ariel_ | asterisk -rx "show channels" |
| 20:13.47 | Curus | Yes, but that doesn't help, that does exactly the same as asterisk -r, show channels, exit would |
| 20:14.53 | Curus | Since it's all done quickly, you have a good chance that nothing else interferes... But also a good chance that something does |
| 20:19.31 | jjshoe | asterisk -rx 'show penis' has a great chance of returning more then you need. if you need something specific use the ami |
| 20:20.30 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 20:21.15 | ariel_ | jjshoe, I understand that, But you can't always go through the ami |
| 20:21.40 | ariel_ | and if they built it iinto asterisk they should have done it correctly |
| 20:22.03 | jjshoe | how can you not go through the ami? |
| 20:23.00 | jjshoe | a missconfigured box or an un-reachable ami doesn't make asterisk -rx the proper solution. |
| 20:24.05 | ariel_ | jjshoe, sometimes you don't have the scripts and other things to be able to go through the ami |
| 20:24.34 | ariel_ | when your at a customer and all they have is a box with asterisk plain jane it's a pain to resolve things |
| 20:24.47 | *** join/#asterisk SirThomas (n=tomc@mail.kendeco.com) |
| 20:25.51 | *** join/#asterisk comprookie2000 (n=david@adsl-065-012-210-216.sip.bct.bellsouth.net) |
| 20:28.06 | jjshoe | pain != right |
| 20:28.08 | jjshoe | but whatever |
| 20:28.25 | jjshoe | you just slap bandaids on your customer's phone system ;) |
| 20:28.58 | ariel_ | no I don't |
| 20:29.16 | ariel_ | but to fix what is wrong is first addon's come after |
| 20:29.45 | outtolunc | was gonna ask if i missed anything, i am guessing not <G> |
| 20:31.04 | ariel_ | jjshoe, seems you are aware of a package you can install quickley to fix things then? |
| 20:31.31 | jjshoe | ariel_ I'm aware of how to use the ami. what problem are you having that you can't seem to resolve quickly? |
| 20:32.11 | jjshoe | <3 IO::Socket |
| 20:34.42 | lesouvage | Does anybody know about a solution that makes it possible to "print" an office document to Asterisk and Asterisk send it away as a fax. |
| 20:34.56 | Qwell | a fax machine |
| 20:35.12 | nightsapper | is back |
| 20:35.39 | ariel_ | jjshoe, seems my 5 plus years I am still not able to develop a quick way to resolve issues that the cli tells you right away.... |
| 20:36.17 | ariel_ | spandsp works 80 to 90 % of time if configured correctly |
| 20:37.33 | jjshoe | I don't see how what you are saying has to relate with getting accurate data back from asterisk -rx vs ami. on any system with any real load you'll get far more then what you expect with -rx |
| 20:37.43 | lesouvage | Qwell: :-) I mean adding a fax option that appears as a printer to choose from and when used Asterisk does all the magic in the background. I would like to skip the printing on paper step. |
| 20:38.09 | denon | oh bother, another fax one |
| 20:38.11 | jjshoe | of course I'm assuming in a programming sense, since the second I see an issue twice I find a self healing way to deal with it *shrug* |
| 20:38.22 | ariel_ | lesouvage, there are addons see voip-info for them that will help |
| 20:39.33 | ariel_ | jjshoe, well I understand and most of time I end up using the ami. But there are times which quick and simple ways are better |
| 20:39.42 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
| 20:39.56 | jjshoe | ariel_ if you say so. |
| 20:40.17 | lesouvage | ariel: I already dd some googling but I will try again. But the idea is not that crazy. I have seen it working almost 20 years ago (not with Asterisk of course, but direct faxing form WP 4.2) |
| 20:41.26 | *** part/#asterisk lirakis_work (n=lirakis@65.200.191.241) |
| 20:42.57 | ariel_ | lesouvage, it works and there are things you can setup for asterisk |
| 20:43.18 | ariel_ | but direct printing to a faxmodem from the pass with wp was easyer |
| 20:43.27 | ariel_ | and direct to an actual anolog line |
| 20:43.40 | *** join/#asterisk baalim (n=baalim@190.199.241.35) |
| 20:43.45 | ariel_ | voip and faxes are not a match made in heaven. |
| 20:43.53 | lesouvage | ariel: the good old WP 4.2 days |
| 20:43.56 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 20:44.22 | ariel_ | right now you can still do the same with efax and other email to fax services |
| 20:44.26 | baalim | hello, i have a basic question about the coding in an asterisk environment. does the coding takes place on the asterisk server or on the actual hardphone |
| 20:45.16 | lesouvage | ariel: That is my idea too but I don't think it is the customers idea. |
| 20:46.04 | errr | lesouvage: we use hylafax and whfc which installs as a printer for the users to "print" to and then using iaxmodem and hylafax it gets sent out though asterisk |
| 20:46.30 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 20:46.37 | Curus | How is your experience with whfc, errr? |
| 20:46.39 | errr | all foss too :) |
| 20:47.08 | errr | Curus: we have been using it here for over 5 years, and a little over 18 months with this asterisk + iaxmodem + hylafax solution |
| 20:47.58 | errr | before we had asterisk we used hylafax + whfc and had regular usb modems going though our intertel system |
| 20:48.36 | Curus | We only offer inbound fax-to-email, but it'd be neat to offer outbound too |
| 20:48.55 | Curus | (Since fax-over-IP isn't really an option with asterisk right now) |
| 20:48.59 | errr | I hate faxes |
| 20:49.09 | ariel_ | so jjshoe, in your view it's easier to telnet to port 5038 then run commands like MI,"Action: Command\r\nCommand: My Fun command\r\n\r\n" |
| 20:49.10 | Nugget | telnet is eeeeeeevil! |
| 20:49.22 | nightsapper | LOL |
| 20:49.29 | ariel_ | argh |
| 20:49.35 | Curus | The fact that asterisk -rx is broken doesn't make it the wrong solution. I just means it should be fixed. |
| 20:49.39 | ariel_ | Nugget, sometimes it is |
| 20:49.50 | Curus | But I'm not volunteering |
| 20:50.58 | ariel_ | I think it's part of this AST_MAX_BUF 256 |
| 20:50.59 | outtolunc | telnet server is evil, telnet client is a MUST |
| 20:51.13 | outtolunc | shakes his head |
| 20:51.27 | Curus | outtolunc: Real men use nc. I prefer telnet though |
| 20:51.27 | *** part/#asterisk RoyK (n=roy@ip-230-7-149-91.dialup.ice.no) |
| 20:52.31 | Curus | And telnetd is an amazing proof that early networking people were completely clueless. Something so simple, and yet you still find security holes 20 years later... |
| 20:52.39 | *** join/#asterisk baalim (n=baalim@190.199.241.35) |
| 20:52.51 | Jumpie | yeah |
| 20:53.01 | Jumpie | internet wasnt designed for the growth it has sustained |
| 20:53.02 | Jumpie | lol |
| 20:53.17 | jjshoe | ariel_ you use telnet? lol. |
| 20:53.27 | ariel_ | some times |
| 20:53.34 | ariel_ | there are so many other ways |
| 20:53.49 | ariel_ | but remember my first post plain jane box |
| 20:54.03 | Curus | They just went for the maximum flexibility... Like the ftp server-to-server mode. |
| 20:54.27 | nightsapper | DARPA was oin a friendly net. No interlopers. So a lot of that stuff was wide open |
| 20:54.47 | Corydon76-dig | asterisk -rx isn't broken. It just doesn't work the way you think it should |
| 20:54.53 | nightsapper | by design - no hostile hackers only firendly ones. |
| 20:55.04 | nightsapper | my how things have changed |
| 20:55.22 | ariel_ | Corydon76-dig, really |
| 20:55.25 | davevg-btwtech | you can write a wrapper script to run the ami commands from the command line, so -rx is not needed |
| 20:55.57 | Curus | Corydon76-dig: What use is the current way it works? |
| 20:55.59 | ariel_ | hello davevg-btwtech, yes your correct. |
| 20:56.00 | jjshoe | Corydon76-dig indeed. |
| 20:56.31 | Corydon76-dig | Curus: The use is to execute CLI commands |
| 20:56.31 | Qwell | Curus: to execute commands |
| 20:56.46 | asdx | [Apr 8 20:56:16] WARNING[26283]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 20:56.47 | Curus | Yes, but why let it output anything at all? |
| 20:56.49 | asdx | what does that means? |
| 20:56.54 | Curus | If you're going to let it output gibberish? |
| 20:57.05 | Qwell | convenience, I suppose |
| 20:57.26 | Curus | At least fix it to not output anything |
| 20:57.32 | Corydon76-dig | In other words, you don't like the output, and you're too lazy to >/dev/null |
| 20:57.42 | Curus | No, I want the result of the command output |
| 20:57.50 | Corydon76-dig | Then use AMI |
| 20:58.01 | Curus | What's wrong with fixing -rx? |
| 20:58.06 | ariel_ | knew that was coming |
| 20:58.08 | Qwell | it's not "fixable" |
| 20:58.14 | Corydon76-dig | The CLI was NEVER intended to be used by automated processes |
| 20:58.25 | outtolunc | fix health care while you are it |
| 20:58.34 | Corydon76-dig | AMI is to be used by processes, not the CLI |
| 20:58.52 | Corydon76-dig | CLI is intended for human operators only |
| 20:58.56 | Curus | It isn't usable by humans either, we get the broken output too |
| 20:59.13 | Corydon76-dig | Curus: we accept patches |
| 20:59.29 | Qwell | the patch will, of course, be about 300k though |
| 20:59.35 | Curus | So we're back to it IS broken, just not fixable. |
| 20:59.59 | Corydon76-dig | Curus: It's not broken. It just doesn't work the way you think it should |
| 21:00.08 | errr | Curus: I use the output of asterisk -rx commands all the time, do you have a problem parsing the data or something? |
| 21:00.10 | ariel_ | wow now that is something |
| 21:00.19 | Curus | errr: Yes, I get random data interspersed |
| 21:00.19 | ariel_ | not broken just does not work like you want it too. |
| 21:00.29 | errr | Curus: using grep? |
| 21:00.45 | Curus | errr: Can't trust grep, the data is too random and can appear in the middle of lines |
| 21:00.56 | ariel_ | Corydon76-dig, sorry but that is a 1/2 answer to no one really wants to fix it |
| 21:01.11 | Curus | Especially if someone else happened to run -rvvvvvvvv |
| 21:01.16 | Corydon76-dig | ariel_: Have you submitted a patch? |
| 21:01.17 | errr | Curus: well I was just going to say we cant read any of the output with grep, I have to use awk to parse out the data we use |
| 21:01.24 | *** join/#asterisk mm_202 (n=mm_202@unaffiliated/mm-202) |
| 21:01.58 | *** join/#asterisk nybbled (n=nybbled@about/apple/performa/nybble) |
| 21:02.09 | Corydon76-dig | This entire discussion comes down to "I want you to patch it." "We don't think it's broken." |
| 21:02.18 | outtolunc | i always love it when people come complaining and do not want to do the work <G> |
| 21:02.22 | ariel_ | Corydon76-dig, no I am not a programmer. And at this stage why would I want ot sign things over to a commercial company |
| 21:02.27 | jjshoe | Corydon76-dig Qwell lots of complaining and no one ponying up money/effort to fix, opensores <3 |
| 21:02.40 | davevg-btwtech | Corydon76-dig, would a cli wrapper for the AMI be something wanted? Should I post it to the bug tracker? |
| 21:02.55 | Curus | Well if it "isn't broken", what are the chances that patches will get through? |
| 21:02.58 | Corydon76-dig | davevg-btwtech: sure, sounds fine |
| 21:03.18 | Corydon76-dig | Curus: Feature adds |
| 21:03.25 | Corydon76-dig | Curus: trunk-only |
| 21:03.34 | jjshoe | davevg-btwtech Corydon76-dig: as in an external program that acts like asterisk -r ? |
| 21:03.37 | outtolunc | patches are for 'changes in functionality' also, especially if they are designed as OPTIONAL |
| 21:04.13 | Qwell | davevg-btwtech: one problem with that, is that you'd need to auth |
| 21:04.13 | Curus | So you think someone would want the current behaviour if the other behaviour was offered? |
| 21:04.33 | davevg-btwtech | got that handled, i'll post the link to the contrib shortly |
| 21:04.37 | outtolunc | wouldn't matter to me, i use the AMI <G> |
| 21:04.44 | *** join/#asterisk adorah (n=Michael@87.69.130.248) |
| 21:04.45 | Corydon76-dig | Things can work as designed (as is the case here) and yet be redesigned to work better. It is not a conflict to say that it wasn't buggy before, and yet something is made better. |
| 21:04.51 | jjshoe | outtolunc same. heh. I have a perl script that acts like asterisk -r to the ami :) |
| 21:05.12 | davevg-btwtech | jjshoe, mine does the same thing :) |
| 21:05.26 | jblack | I'm checking up on the R1T1. The phone company is asking me what protocol I need with my pri... DMS100, 4ESS, N12, 5ESS9 and 5ESS10. |
| 21:05.29 | jjshoe | davevg-btwtech <3 |
| 21:05.53 | Corydon76-dig | ariel_: You probably need to sit down with your lawyer to realize that changes aren't actually signed over to us |
| 21:05.55 | jblack | How can I learn how to answer that question in a non-"huh?" type manner? |
| 21:06.12 | jjshoe | jblack 4ess is fine |
| 21:06.25 | jjshoe | so is dms100 |
| 21:06.56 | *** join/#asterisk saftsack (n=oliver@p4FC75C46.dip.t-dialin.net) |
| 21:07.03 | *** join/#asterisk b1ch0 (n=ralabiso@190.129.29.49) |
| 21:07.05 | Corydon76-dig | Submitted patches are licensed which is a completely different legal principle |
| 21:07.07 | ariel_ | Corydon76-dig, really.... hummm I have. |
| 21:07.35 | ariel_ | That is the issue with them the dual license |
| 21:07.39 | Corydon76-dig | ariel_: a competant lawyer? |
| 21:07.45 | ariel_ | yes |
| 21:08.10 | Corydon76-dig | If he told you you were giving up your rights to the code, he clearly wasn't |
| 21:08.12 | jblack | It seems as if these pri protocols are designed for specific hardware, and the rhino cards essentially emulates a variety of them? |
| 21:08.14 | jjshoe | jblack it's a switch type. |
| 21:08.25 | Jumpie | hi jjshoe :) |
| 21:08.26 | jjshoe | jblack you might want to grab the sample zapata |
| 21:08.29 | jjshoe | hello Jumpie |
| 21:08.43 | jjshoe | jblack http://www.voip-info.org/wiki/index.php?page=Asterisk+config+zapata.conf.sample |
| 21:09.13 | Jumpie | is it a big pain in the butt to get asterisk downlaoded and compiled |
| 21:09.18 | jjshoe | Jumpie no. |
| 21:09.19 | Jumpie | and then LATER get a zap card |
| 21:09.27 | jjshoe | Jumpie no. |
| 21:09.29 | Jumpie | k |
| 21:09.32 | davevg-btwtech | http://bugs.digium.com/view.php?id=12389 |
| 21:10.12 | Qwell | davevg-btwtech: neat |
| 21:10.41 | Curus | Thank you davevg-btwtech |
| 21:11.01 | jblack | Hmmm. if I'm defining switchtype within *, then that would imply to me that switchtype isn't a function of a T1 card like the rhino or digium cards, but of asterisk. Is that correct? |
| 21:11.45 | jjshoe | davevg-btwtech I use IO::Socket :) |
| 21:11.56 | davevg-btwtech | either way works :) |
| 21:12.00 | jjshoe | indeed |
| 21:12.12 | *** join/#asterisk aksyn (n=aksyn@78.86.127.226) |
| 21:12.21 | davevg-btwtech | feel free to make it work in more cases so it can work for a larger audience |
| 21:12.21 | aksyn | how does asterisk compare to this? http://digg.com/software/An_Open_Source_Class_Five_Telephony_Switch |
| 21:12.31 | jjshoe | the only issue is I don't believe Net::Telnet is part of the standard perl distribution. |
| 21:12.43 | Qwell | jjshoe: it's not.. I'm having to cpan it |
| 21:13.30 | jjshoe | let me email myself the bug thingy and I'll whip something up tonight |
| 21:13.45 | Qwell | You're gonna have to get Fonality to let you sign the license agreement |
| 21:14.05 | *** join/#asterisk jmhunter (n=jmhunter@72.14.86.68) |
| 21:14.05 | *** mode/#asterisk [+o jmhunter] by ChanServ |
| 21:14.07 | jblack | I guess to answer my question more clearly, is how do I know some particular card can be confused by asterisk to use 4ESS, DMS100, etc. |
| 21:14.21 | jjshoe | jblack configured? |
| 21:14.27 | jjshoe | Qwell hrm, I'll see what happens :P |
| 21:14.35 | jblack | gah. freudian slip. :) |
| 21:14.36 | davevg-btwtech | i could re-write it to use io::socket if jjshoe can't get his stuff disclaimed |
| 21:14.53 | file | aksyn: well Asterisk was originally designed as a PBX, not a switch... so comparing the two is like comparing an apple to an orange |
| 21:14.53 | jjshoe | davevg-btwtech yeah, use IO::Socket, additionally you need to support -r, and -x |
| 21:14.55 | jjshoe | or -rx |
| 21:14.56 | jblack | I guess my question, more clearly asked, is how do I know some particular card can be configured by asterisk to use 4ESS, DMS100, etc. |
| 21:14.57 | jjshoe | etc |
| 21:15.05 | jjshoe | jblack buy a card that doesn't suck |
| 21:15.05 | aksyn | file: can't FS be a pbx too? |
| 21:15.06 | jmhunter | bueno |
| 21:15.16 | Qwell | I don't think -rx is needed. it'd be implied by using astcli vs asterisk |
| 21:15.21 | file | jblack: that is protocol level, implemented by libpri |
| 21:15.31 | *** part/#asterisk jmhunter (n=jmhunter@72.14.86.68) |
| 21:15.34 | Qwell | maybe ignoring them isn't a bad idea though *shrug* |
| 21:15.41 | file | aksyn: maybe, I don't know... |
| 21:16.02 | *** join/#asterisk lanwifie (n=lanwifie@75-105-17-160.cust.wildblue.net) |
| 21:16.28 | *** join/#asterisk _-Jon-_ (n=jonmiron@CPE000d8861e8f7-CM00080d290642.cpe.net.cable.rogers.com) |
| 21:16.39 | jjshoe | Qwell eh, i guess it's up to you guys, but I would want the same behaviour as "asterisk" |
| 21:16.43 | _-Jon-_ | Hello all |
| 21:17.09 | *** join/#asterisk jmhunter (n=jmhunter@72.14.86.68) |
| 21:17.09 | *** mode/#asterisk [+o jmhunter] by ChanServ |
| 21:17.44 | _-Jon-_ | I'm having a codec issue, and wondering if someone can clear this up. I have a Sipura and Grandstream phone, both configured to use G729, all seems well when I call, however if I try to hit # to transfer, it hangs up because of a translation problem |
| 21:17.49 | file | aksyn: capabilities of FS would be best discussed in #freeswitch with people who have experience with it |
| 21:17.56 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 21:17.56 | *** mode/#asterisk [+o anthm] by ChanServ |
| 21:18.01 | aksyn | file: ack, am in there |
| 21:18.02 | nightsapper | G729 DTMF in band = no go |
| 21:18.33 | nightsapper | usually too compressed for accuate DTMF, at least in my limited experience |
| 21:18.51 | _-Jon-_ | Hmmm so I shouldn't bother with G729? |
| 21:18.55 | jjshoe | Qwell davevg-btwtech additionally, as I have never looked into it, can the ami be configured to use different interfaces to listen on? perhaps the config should take in an ip/host... |
| 21:19.33 | nightsapper | well... depends on why you need it. and if you can figure how to send the DTMF OOB. Im not an expert. One of the braniacs here may know |
| 21:19.37 | davevg-btwtech | could do that, default to 5038 unless overwritten |
| 21:19.52 | jjshoe | hrm port is good too :P |
| 21:19.56 | _-Jon-_ | I just figured I'd use it for bandwidth saving reasons :P |
| 21:19.57 | Qwell | davevg-btwtech: hostname too - it needn't be localhost |
| 21:19.58 | Curus | Grandstreams generally need DTMF set to SIP INFO, all other phones are generally better off with RFC<mumble> |
| 21:20.12 | davevg-btwtech | ok, i'll see what i can whip up tomorrow :) |
| 21:20.13 | Qwell | (though, default to localhost:5038, of course) |
| 21:20.16 | nightsapper | RFC<3evil> |
| 21:20.25 | _-Jon-_ | I guess really what I want to know is how I can resolve a message like this: [Apr 8 17:18:33] WARNING[5032]: file.c:199 ast_writestream: Unable to translate to format gsm, source format g729 |
| 21:20.36 | nightsapper | Oh you want transcoding |
| 21:21.16 | _-Jon-_ | I guess so :P |
| 21:21.31 | _-Jon-_ | All I did was disallow=all and allow=g729 in sip.conf for the 2 phones |
| 21:21.51 | nightsapper | Not quite sure how to set that up. I usually use GSM or G711 for everything. might add allow GSM in there |
| 21:21.51 | file | it's trying to write audio out to a file. |
| 21:22.12 | nightsapper | VM? |
| 21:22.23 | _-Jon-_ | Ohhh I forgot, it records all the calls |
| 21:22.30 | nightsapper | and be sure you have alw and mlaw as well |
| 21:22.48 | nightsapper | thats what it usually uses for call recording. |
| 21:22.57 | lanwifie | http://digg.com/software/An_Open_Source_Class_Five_Telephony_Switch |
| 21:22.58 | nightsapper | *alaw. |
| 21:22.58 | *** part/#asterisk lanwifie (n=lanwifie@75-105-17-160.cust.wildblue.net) |
| 21:23.37 | _-Jon-_ | Where can I get alw and mlaw from? |
| 21:24.44 | ariel_ | davevg-btwtech, hope everything is going well, I should be back in the office Thursday. Still have not seen the list from you I asked for... |
| 21:26.22 | davevg-btwtech | ariel_, we'll catch up tomorrow |
| 21:28.49 | ariel_ | night folks.... been fun and long time since I have been in here. |
| 21:29.07 | *** join/#asterisk Linker3000L (n=chatzill@78.32.25.201) |
| 21:29.30 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:30.28 | *** join/#asterisk branchcut (n=tleyden@adsl-69-111-167-235.dsl.pltn13.pacbell.net) |
| 21:31.52 | *** part/#asterisk khaefner (n=khaefner@129.82.174.114) |
| 21:32.14 | *** join/#asterisk cootoa (n=cootoa@maa78-2-82-241-93-36.fbx.proxad.net) |
| 21:32.23 | _-Jon-_ | Okay now I have another question, about parked calls.. Do I need to modify extensions.conf at all, or just features.conf? |
| 21:32.31 | cootoa | hi |
| 21:32.34 | *** join/#asterisk asdx (n=diego@adsl-157-61.click.com.py) |
| 21:34.10 | [TK]D-Fender | _-Jon-_, you need to include the [parkedcalls] context in the context used by your phones, and the rest is features.conf |
| 21:34.31 | _-Jon-_ | Oh, I don't have a parkedcalls context |
| 21:35.13 | Corydon76-dig | davevg-btwtech: added, thank you |
| 21:35.19 | [TK]D-Fender | _-Jon-_, Its created automatically. |
| 21:36.26 | _-Jon-_ | Ohhh okay, thanks :) |
| 21:36.52 | Corydon76-dig | davevg-btwtech: oh, and, I made a tiny change |
| 21:37.10 | *** join/#asterisk troubled_ (n=troubled@pdpc/supporter/sustaining/troubled) |
| 21:37.33 | *** join/#asterisk dmz (n=dmz@adsl-99-163-4-133.dsl.pltn13.sbcglobal.net) |
| 21:37.33 | _-Jon-_ | Okay and one FINAL question :) |
| 21:37.42 | jjshoe | Corydon76-dig that thing needs quite a bit of work :P |
| 21:37.47 | cootoa | what file should i modify, I have 2 ipbx: ipbx1 that has external call setup right, and ipbx2 |
| 21:38.03 | Corydon76-dig | jjshoe: what are you talking about? |
| 21:38.30 | jjshoe | Corydon76-dig support for asterisk on a diff interface/port at the very least. |
| 21:38.40 | _-Jon-_ | I have 2 phones (a sipura ATA and a grandstream phone), when I call out using the ATA I can press # to transfer (or park), but on the Grandstream # just transmit the tone. Am I missing something, or is that just the way it is? |
| 21:38.49 | cootoa | i setup the iax trunk everything works fine but i dont know what to do so ipphone from ipbx2 can make external call through ipbx1 |
| 21:38.56 | *** part/#asterisk branchcut (n=tleyden@adsl-69-111-167-235.dsl.pltn13.pacbell.net) |
| 21:38.59 | jjshoe | Corydon76-dig that and Net::Telnet isn't in standard perl distrib. I'd let him convert it to IO::Socket. |
| 21:39.26 | [TK]D-Fender | _-Jon-_, Ok, you are using 2 devices that each have their own native way of transferring calls that doesn't involve using *'s DTMF transfers via "#". use them instead. |
| 21:39.35 | Corydon76-dig | jjshoe: already committed to trunk |
| 21:39.41 | *** join/#asterisk mindCrime (n=chatzill@64.241.37.140) |
| 21:39.41 | jjshoe | ouch. |
| 21:39.53 | [TK]D-Fender | _-Jon-_, a lot of the docs are BS in making it sound like you need to use the "tT" dial features for parking, etc. |
| 21:40.02 | Corydon76-dig | jjshoe: there's nothing wrong with it |
| 21:40.14 | Corydon76-dig | jjshoe: perhaps a few features to be added... |
| 21:40.14 | jjshoe | Corydon76-dig I don't disagree. |
| 21:40.50 | _-Jon-_ | You're right, the native transfer button does work on the phone itself, but when I transfer to 700 to park it, it doesn't read back the parked extension which is a bit of a problem, hence why I wanted to use # instead |
| 21:41.06 | asdx | i was using talking with some guy over SIP with asterisk, the other end point was a softphone, i was using a softphone too... but i heard some echo/fuzzy-noise, i was using the gsm codec |
| 21:41.08 | _-Jon-_ | D-Fender, I was always under that impression that I needed it |
| 21:41.28 | asdx | how can i improve that? |
| 21:42.38 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 21:43.18 | [TK]D-Fender | _-Jon-_, You don't need it. But what you DO need to do is make sure that you're doing an ATTENDED transfer to the parking lot, and not a BLIND transfer. |
| 21:43.49 | [TK]D-Fender | asdx, get a beater headset |
| 21:43.51 | _-Jon-_ | D-Fender, ohhh okay so what I'm doing on my GS must be a blind transfer |
| 21:44.04 | [TK]D-Fender | _-Jon-_, thats what I'd guess. |
| 21:44.55 | _-Jon-_ | D-Fender, hmm, I like the attended transfer, however some numbers I dial I need to press #. guess I can't have it both ways |
| 21:45.00 | asdx | <PROTECTED> |
| 21:45.01 | asdx | <PROTECTED> |
| 21:45.29 | [TK]D-Fender | asdx, Try another code then. |
| 21:46.03 | cootoa | someone has an idea? |
| 21:46.12 | asdx | [TK]D-Fender: internet is crappy in Paraguay, maybe is that |
| 21:46.26 | [TK]D-Fender | _-Jon-_, Sure you can. Don't use "t" or "T" in your Dial call, and tell your phone not to mess with that in its dialplan either |
| 21:46.34 | [TK]D-Fender | asdx, maybe |
| 21:46.48 | _-Jon-_ | D-Fender, but I don't suppose I can do this with an ATA can I? |
| 21:47.11 | [TK]D-Fender | _-Jon-_, Sure, either of them can. |
| 21:47.46 | _-Jon-_ | D-Fender, do enlighten me :D I'd love to be able to do this, but I can't see any other way other than pressing # on my phone |
| 21:47.52 | [TK]D-Fender | _-Jon-_, Sipura ATA's work just great via hook-flash transfers, and the GS phone you should be able to do that via dedicated hard buttons easily |
| 21:48.13 | _-Jon-_ | D-Fender you my friend area genius :) |
| 21:48.16 | _-Jon-_ | *are |
| 21:48.31 | [TK]D-Fender | cootoa, Go look up "asterisk dual servers" on the wiki. |
| 21:48.54 | cootoa | 2 ipbx, ipbx1 makes external call ipbx2 & ipbx1 have an iax trunk they can call each other what should i do to make ipbx2 to make external through ipbx1? |
| 21:49.01 | [TK]D-Fender | _-Jon-_, glad to help. |
| 21:49.21 | [TK]D-Fender | cootoa, I just told you where to go to lean about how to hook your servers up. Get to it. |
| 21:49.49 | _-Jon-_ | :) |
| 21:50.03 | cootoa | o didnt see :p |
| 21:50.25 | cootoa | thanks |
| 21:52.07 | *** join/#asterisk jmhunter (n=jmhunter@72.14.86.68) |
| 21:52.07 | *** mode/#asterisk [+o jmhunter] by ChanServ |
| 21:52.15 | asdx | hm nice |
| 21:52.21 | asdx | my telco doesn't block SIP packets anymore |
| 21:53.38 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 21:54.59 | _-Jon-_ | your isp? |
| 21:55.06 | saftsack | i hate the patton m-ata pin alignment ... my fax doesnt work with a standard cable,humm |
| 21:55.32 | asdx | _-Jon-_: yes |
| 21:55.48 | _-Jon-_ | asdx, wow that's just evil |
| 21:56.35 | [hC] | Anyone here used Aastra phones on a Cisco PoE switch? They seem to want to continuously reboot instead of properly starting up |
| 21:56.48 | lesouvage | Does any of you use the G7.22 codec (wide band audio) for internal phone calls and does it really makes a difference? |
| 21:56.52 | asdx | _-Jon-_: yeah |
| 21:57.01 | asdx | [TK]D-Fender: i'm running * on a vps |
| 21:57.18 | _-Jon-_ | Good think my VoIP traffic goes through a VPN, so until they block port 1194 I'm okay :P |
| 21:57.35 | asdx | [TK]D-Fender: can that be the problem of the audio? |
| 21:58.11 | [TK]D-Fender | asdx, if the packets are flowing, its source audio issues, or jitter at one end or the other |
| 21:58.31 | *** join/#asterisk SteveTotaro (n=Administ@pool-151-196-236-69.balt.east.verizon.net) |
| 21:58.46 | SteveTotaro | whats up all? |
| 21:59.02 | outtolunc | party at your friends place!! (jk'n) |
| 21:59.12 | _-Jon-_ | lol |
| 21:59.36 | outtolunc | let him know he can adopt me <G> |
| 21:59.47 | asdx | [TK]D-Fender: i see, thanks |
| 22:03.20 | Corydon76-dig | Adopt-A-Marine ? |
| 22:03.36 | outtolunc | with all those toys, yep |
| 22:03.48 | outtolunc | an allowance would be nice also <G> |
| 22:07.06 | cootoa | dual server isnt really what I am looking for [TK]D-Fender |
| 22:07.54 | *** join/#asterisk jmhunter (n=jmhunter@72.14.86.68) |
| 22:07.54 | *** mode/#asterisk [+o jmhunter] by ChanServ |
| 22:08.38 | *** join/#asterisk [intra]lanman (n=lanman@va-76-6-215-38.dhcp.embarqhsd.net) |
| 22:08.58 | cootoa | I am tired i'll come back tomorow |
| 22:10.25 | *** join/#asterisk marc7 (n=marc@64.46.14.64.novuscom.net) |
| 22:10.57 | marc7 | i'm trying to import a conference .ulaw file into audacity for playback/editing... but I'm having trouble finding the settings I need to import the raw data in. any suggestions? |
| 22:14.50 | *** join/#asterisk SallySegFault (n=lanwifie@75-105-17-160.cust.wildblue.net) |
| 22:15.37 | SallySegFault | i just thought i belonged here :) |
| 22:15.41 | *** join/#asterisk jmhunter (n=jmhunter@72.14.86.68) |
| 22:15.41 | *** mode/#asterisk [+o jmhunter] by ChanServ |
| 22:15.43 | SallySegFault | smiles innocently |
| 22:15.54 | *** join/#asterisk bkw__ (n=brian@adsl-70-234-166-37.dsl.tul2ok.sbcglobal.net) |
| 22:15.55 | [intra]lanman | SallySegFault: i think you do too |
| 22:16.05 | SallySegFault | why thank you |
| 22:16.38 | [intra]lanman | SallySegFault: what's your favorite part of asterisk? |
| 22:16.46 | [intra]lanman | the deadlocks or the segfaults? |
| 22:17.07 | outtolunc | the delay caused by the sat link? |
| 22:17.15 | SallySegFault | segfaults of course :) i mean it's what i'm all about :) |
| 22:17.19 | *** mode/#asterisk [+b *!n=lanwifie@*.cust.wildblue.net] by Qwell |
| 22:17.19 | *** kick/#asterisk [SallySegFault!i=north@pdpc/sponsor/digium/Qwell] by Qwell (Qwell) |
| 22:17.22 | *** mode/#asterisk [+b *!*n=lanman@*.dhcp.embarqhsd.net] by Qwell |
| 22:17.22 | *** kick/#asterisk [[intra]lanman!i=north@pdpc/sponsor/digium/Qwell] by Qwell (Qwell) |
| 22:17.34 | Qwell | moving right along |
| 22:17.39 | outtolunc | hehe |
| 22:18.11 | bkw__ | Qwell: how you doing today? |
| 22:18.22 | Qwell | doing great |
| 22:18.45 | bkw__ | thats good to hear... what's new with you these days? |
| 22:18.54 | bkw__ | I see that their is a free switchvox version.. thats a plus I suspect |
| 22:19.35 | bkw__ | how long ago did the switchvox free verison come out? |
| 22:19.39 | aksyn | actually on that note, is there anything out there that adds resilience onto asterisk? (not an openser solution, i'm thinking something that can keep active calls up) |
| 22:19.50 | Qwell | bkw__: dunno, 2-3 months ago? |
| 22:20.07 | bkw__ | odd I didn't see anything hit any news site... guess I wasn't paying attention |
| 22:20.19 | bkw__ | I guess it covers the lower end installs you'll never make money on anyway |
| 22:21.00 | asdx | [TK]D-Fender: they say the quality is good with skype, so can it still be their headphones? |
| 22:21.47 | asdx | [TK]D-Fender: maybe is just codecs? |
| 22:22.13 | bkw__ | asdx: what codecs are they trying to use? and what does it sound like? |
| 22:23.01 | *** join/#asterisk ManxPower (n=manxpowe@57.sub-70-221-144.myvzw.com) |
| 22:24.36 | asdx | bkw__: it sounds faded |
| 22:24.44 | bkw__ | what codec? |
| 22:25.12 | asdx | bkw__: there was gsm, ulaw and ilbc on sip.conf |
| 22:25.21 | bkw_ | all of them fade? is this sip to sip? |
| 22:25.23 | bkw_ | or sip to tdm? |
| 22:25.29 | asdx | sip to sip |
| 22:25.36 | bkw_ | what phones? |
| 22:25.43 | asdx | zoiper on both endpoints |
| 22:25.54 | asdx | i will try with gsm now |
| 22:26.01 | bkw_ | I have seen some similar issues with Zoiper when I was tinkering with it on FreeSWITCH |
| 22:26.10 | bkw_ | but I think the latest biz release I have is fine now |
| 22:26.22 | bkw_ | I don't hear the strangeness anymore |
| 22:26.47 | asdx | bkw_: they are using the free version of zoiper |
| 22:27.03 | bkw_ | I had bad luck getting that working on my mac |
| 22:27.11 | bkw_ | you using winders? |
| 22:27.18 | asdx | bkw_: they are on windows |
| 22:27.19 | asdx | i use linux |
| 22:27.30 | bkw_ | hrm let me get the free one again and see |
| 22:28.23 | *** join/#asterisk talntidwrk (n=t@66.208.251.170) |
| 22:28.24 | asdx | gsm is a good codec right |
| 22:29.04 | bkw_ | well depends |
| 22:29.08 | bkw_ | their are some that are broken |
| 22:29.17 | bkw_ | the one in Asterisk I would classify as broken |
| 22:29.20 | bkw_ | it works |
| 22:29.22 | bkw_ | don't get me wrong |
| 22:29.27 | bkw_ | it won't pass the ITU specs |
| 22:29.31 | bkw_ | or ETSI |
| 22:29.33 | DrkShadow | how do you guys assure 99.999% uptime? what do you do for database storage? asterisk server? other?.. |
| 22:29.48 | bkw_ | DrkShadow: their is no such thing as 5 nines with Asterisk |
| 22:29.53 | bkw_ | you can cluster |
| 22:29.54 | asdx | bkw_: but they work well right |
| 22:30.00 | bkw_ | asdx: yes |
| 22:30.00 | asdx | bkw_: usually |
| 22:30.24 | bkw_ | DrkShadow: connecting your DB directly to your switch isn't usually a wise thing |
| 22:30.26 | DrkShadow | bkw: I don't see why not, given my experience |
| 22:30.49 | asdx | bkw_: i told them to try x-lite |
| 22:30.57 | bkw_ | asdx: that thats almost as bad |
| 22:31.06 | *** join/#asterisk Katty (n=The@adsl-68-92-251-223.dsl.stlsmo.swbell.net) |
| 22:31.19 | asdx | shit |
| 22:31.32 | Katty | really. |
| 22:32.10 | bkw_ | x-lite? |
| 22:33.10 | *** join/#asterisk jmhunter (n=jmhunter@72.14.86.68) |
| 22:33.10 | *** mode/#asterisk [+o jmhunter] by ChanServ |
| 22:34.32 | jmhunter | hey brian |
| 22:34.36 | jmhunter | bkw_ |
| 22:34.47 | jmhunter | long time no chat |
| 22:34.52 | bkw_ | hey guy |
| 22:34.53 | bkw_ | what up? |
| 22:34.57 | jmhunter | nada |
| 22:35.04 | [hC] | woop |
| 22:35.13 | bkw_ | its [hC] too |
| 22:35.24 | [hC] | does a little dance |
| 22:35.27 | [hC] | sup brian :) |
| 22:35.37 | bkw_ | not much |
| 22:35.38 | bkw_ | you? |
| 22:36.25 | [hC] | Workin away, workin away :) |
| 22:36.57 | [hC] | Watched a demo on aastra's new embedded pbx today.. I cant wait for their "open" version to come out. The audio ports and remote relay for door strikes is pretty nifty. |
| 22:37.24 | asdx | codec appears as UNKNOWN in zoiper |
| 22:37.29 | bkw_ | weird |
| 22:37.30 | bkw_ | sounds like a bug |
| 22:37.46 | *** join/#asterisk SteveTotaro (n=Administ@pool-151-196-236-69.balt.east.verizon.net) |
| 22:41.25 | *** join/#asterisk RoyK (n=roy@ip-253-58-149-91.dialup.ice.no) |
| 22:41.52 | asdx | <PROTECTED> |
| 22:43.08 | bkw_ | that message could mean two things |
| 22:43.12 | bkw_ | codec or bad transport |
| 22:43.17 | bkw_ | IP4 vs IP6 |
| 22:44.04 | asdx | how do i check which one is ipv4/ipv6 |
| 22:44.13 | bkw_ | doubt thats it |
| 22:44.38 | asdx | it's the codecs probably |
| 22:44.46 | bkw_ | yah could be |
| 22:45.47 | *** join/#asterisk jmesquita (n=jmesquit@200.170.114.149) |
| 22:47.55 | asdx | i don't know why i get Codec: UNKNOWN in zoiper though |
| 22:48.13 | asdx | may be a bug in zoiper |
| 22:48.15 | asdx | i will try xlite |
| 22:49.57 | *** join/#asterisk Olobola (i=Olobola@26.sub-75-209-152.myvzw.com) |
| 22:50.54 | asdx | what happens if you have like 2 or 3 codecs in your sip.conf |
| 22:51.12 | asdx | how the phone knows which one to use? |
| 22:51.31 | bkw_ | ok that is selected via a few methods |
| 22:51.35 | bkw_ | the proper way is |
| 22:51.41 | bkw_ | see the SDP has your RTP map |
| 22:51.49 | bkw_ | me calling you giving you my rtp map.. you respond with yours |
| 22:52.00 | bkw_ | now I can send you any code in that map at any time and you are to deal with it. |
| 22:52.15 | bkw_ | many things dont do this .. they 200OK with only one SINGLE codec in the sdp |
| 22:52.20 | bkw_ | which is the clue on what codec to use |
| 22:52.31 | Olobola | I'm trying to register with two different clients, one works the other fails for some reason. Is it possible to tell by looking at sip headers? |
| 22:52.37 | Olobola | http://www.pastebin.ca/977587 |
| 22:52.49 | asdx | bkw_: oh, so if your phone supports codec 'foo' the server also have to support that codec and my phone as well? |
| 22:52.53 | Olobola | I seperated failed and successful by ################### |
| 22:53.07 | bkw_ | asdx: in this case yes... unless you're going via a proxy |
| 22:53.37 | bkw_ | Olobola: the username and passwords are wrong |
| 22:53.39 | bkw_ | or are you behind nat? |
| 22:53.51 | bkw_ | oh you're using diebeam |
| 22:53.51 | Olobola | bkw_: no nat |
| 22:53.57 | bkw_ | that is the worst softphone on the planet |
| 22:54.23 | Olobola | bkw_: eyebeam works, veronix activex (openwengo) does not |
| 22:54.33 | bkw_ | openwengo is ass too |
| 22:54.38 | bkw_ | its by far worthless |
| 22:54.47 | bkw_ | does nothing but crash on my mac |
| 22:54.53 | bkw_ | even the dev build won't make phones calls |
| 22:55.00 | Olobola | ouch |
| 22:55.13 | bkw_ | Zoiper is ok |
| 22:55.19 | bkw_ | its better than diebeam |
| 22:55.19 | Olobola | I need an activex phone that doesn't cost 10k |
| 22:55.48 | bkw_ | Olobola: does it have to be SIP? |
| 22:56.03 | bkw_ | Olobola: I know someone that has a java IAX one if you wanna email me your details I can forward it over to him |
| 22:59.03 | husimon | mm my polycom soundstation 4000's came in today :) |
| 22:59.09 | husimon | GOODBYE CISCO 7935s! |
| 22:59.16 | Qwell | husimon: send em' here |
| 22:59.21 | Qwell | I'll gladly take them :p |
| 22:59.33 | husimon | what are you going to use them for? |
| 22:59.47 | bkw_ | I like my snom phones |
| 22:59.47 | Qwell | to play with. I like skinny phones |
| 22:59.59 | husimon | The friggin soft buttons don't work and they are flakey |
| 23:00.07 | bkw_ | bbl |
| 23:00.07 | *** part/#asterisk bkw_ (n=brian@adsl-70-234-166-37.dsl.tul2ok.sbcglobal.net) |
| 23:00.20 | Qwell | they don't work with chan_skinny because I've never had one :D |
| 23:00.34 | Qwell | flakey...yeah, that doesn't surprise me |
| 23:01.28 | husimon | you have a call manager? |
| 23:01.32 | Qwell | nope |
| 23:01.41 | husimon | what are you using? chan_sccp? |
| 23:01.50 | Qwell | I'm the maintainer of chan_skinny |
| 23:01.56 | husimon | ah oh, now I get it :) |
| 23:02.07 | husimon | I'll tell you what, you pay shipping and I'll send you one. |
| 23:02.13 | Qwell | really? |
| 23:02.29 | husimon | Ya they are just going to sit here and rot |
| 23:02.32 | husimon | unless I ebay them |
| 23:02.33 | Qwell | you in the US? |
| 23:02.36 | husimon | i'm in hawaii |
| 23:02.51 | Qwell | so basically no :) but no import fees. hmm |
| 23:03.07 | husimon | probably pretty minimal shipping costs though |
| 23:03.20 | Qwell | never shipped to/from HI. |
| 23:04.07 | husimon | what's your zip, i'll try and see what the costs are |
| 23:05.40 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 23:06.24 | asdx | where do you put the hostname in x-lite? |
| 23:07.17 | asdx | nvm, found it |
| 23:18.11 | *** join/#asterisk lirakis (n=lirakis@cpe-68-175-38-65.nyc.res.rr.com) |
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| 23:20.06 | *** join/#asterisk gmaruz1 (n=maruzz@213-140-6-122.ip.fastwebnet.it) |
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| 23:26.13 | jblack | sighs |
| 23:26.53 | jblack | Give someone xen, lvm, ldap, samba, ltsp, their own nameservers, proxy servers, asterisk... everything one could want.. And what does the company do with it? |
| 23:27.41 | jblack | They share mp3s. |
| 23:32.53 | *** join/#asterisk tobias (n=tobias@user-0ce2hpk.cable.mindspring.com) |
| 23:35.02 | asdx | jblack: lol |
| 23:35.45 | jblack | Yeah. 200 hours of my work so that they could become the great Tenacious D repo. |
| 23:35.58 | asdx | you are joking right? |
| 23:36.23 | jblack | shakes his head sadly |
| 23:36.35 | asdx | :S |
| 23:36.57 | jblack | That's why I had to bump the "IT admin"'s quota up from 500 megs to 10 gigs. |
| 23:37.06 | asdx | always see who are your customers, i guess |
| 23:37.36 | jblack | well... on the bright side... if they ever fail to pay, I know where the BSA and RIAA are. ;) |
| 23:37.47 | asdx | heh |
| 23:38.37 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 23:39.52 | *** part/#asterisk mm_202 (n=mm_202@unaffiliated/mm-202) |
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| 23:42.59 | Olobola | I'm trying to register with two different clients, one works the other fails for some reason. Is it possible to tell by looking at sip headers? |
| 23:43.02 | Olobola | I need an activex phone that doesn't cost 10k |
| 23:43.12 | Olobola | whoops |
| 23:43.21 | Olobola | http://www.pastebin.ca/977587 |
| 23:43.27 | Olobola | I seperated failed and successful by ################### |
| 23:45.32 | *** join/#asterisk SteveTotaro (n=Administ@pool-151-196-236-69.balt.east.verizon.net) |
| 23:46.37 | *** join/#asterisk jmhunter (n=jmhunter@adsl-71-149-242-195.dsl.mtry01.sbcglobal.net) |
| 23:46.37 | *** mode/#asterisk [+o jmhunter] by ChanServ |
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| 23:48.24 | *** mode/#asterisk [+o jmhunter] by ChanServ |
| 23:56.04 | *** join/#asterisk Keizer (n=keizer@208.50.100.60) |
| 23:56.09 | Keizer | Sup |
| 23:56.20 | JayTee52 | I'm pondering |
| 23:57.00 | Keizer | I'm testing out Asterisk 1.6 |
| 23:57.06 | JayTee52 | how is it? |
| 23:57.09 | Keizer | About to try an IAX setup with virtual machines |
| 23:57.14 | JayTee52 | cool |
| 23:57.23 | Keizer | So far my CISCO 7960 seems bugged a little |
| 23:57.40 | Keizer | I can make calls from it and to it but no audio goes through and then the call drops |
| 23:57.44 | JayTee52 | I'm waiting awhile till it's more stable to test it, too busy to get into the beta at this point. |
| 23:58.15 | JayTee52 | are you using skinny? |
| 23:58.53 | Keizer | Using SIP |
| 23:59.38 | JayTee52 | so the Cisco's support SIP? I thought they used skinny by default |
| 23:59.53 | ManxPower | JayTee52: not all ciscos do and yes, they come with skinny |
