00:07.54 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-91-250.cybersurf.com) |
00:12.55 | *** join/#asterisk Gabriel25 (n=whatever@user-12ld5f7.cable.mindspring.com) |
00:13.57 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
00:16.35 | djacob-wcgs | hello all, how are you , i have a question, im trying to send calls from eyebeam sip phone --->sip ---> asterisk--->sip --->cisco 5400---isdn -->pstn and then i call out the pstn to a pbx , it does not detect dtmf tones |
00:16.50 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
00:16.56 | djacob-wcgs | can anyone point me in the right direction for some reading about this |
00:17.15 | djacob-wcgs | or have any ideas, im just learning all this |
00:17.18 | djacob-wcgs | thanks in advance :) |
00:17.20 | *** join/#asterisk zotz (n=zotz@24.244.133.115) |
00:18.17 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
00:18.17 | *** mode/#asterisk [+o denon] by ChanServ |
00:19.13 | djacob-wcgs | my codec is g7.11 also |
00:20.41 | Gabriel25 | guys IF I have a regular IP phone .... no sip |
00:20.54 | Gabriel25 | how I can register with my asterisk box ? |
00:21.14 | Gabriel25 | I have avaya 4624 ip phone |
00:21.48 | Gabriel25 | R1.8.3 H.323 release for the 4606/4612/4624 I update with this version |
00:22.46 | Gabriel25 | How i can make this IP phone working with my asterisk box ? |
00:23.54 | *** join/#asterisk fiXXXerMet (n=Kyle@cmu-24-35-80-91.mivlmd.cablespeed.com) |
00:25.18 | [TK]D-Fender | Gabriel25 : Did you setup H.323 on *? |
00:25.52 | Gabriel25 | I don`t think so |
00:26.10 | Gabriel25 | How I can check that |
00:28.00 | *** part/#asterisk Lord_Drachenblut (n=Lord@12.210.100.18) |
00:28.04 | [TK]D-Fender | Gabriel25 : Well that phone runs on H.323... I think you'd better do a whole lot more reading first.... the phone isn't the problem yet. |
00:28.31 | Gabriel25 | I know I have to read a lot but I need a start ! |
00:28.39 | *** part/#asterisk fiXXXerMet (n=Kyle@cmu-24-35-80-91.mivlmd.cablespeed.com) |
00:28.54 | Gabriel25 | I don`t know if I asterisk support H.323 |
00:29.14 | Gabriel25 | or if are exist ..... a sip firmware for this phone |
00:29.51 | *** join/#asterisk robl^ (n=robl@dsl093-025-218.hou1.dsl.speakeasy.net) |
00:29.59 | Gabriel25 | http://support.avaya.com/japple/css/japple?temp.documentID=283920&temp.productID=107755&temp.bucketID=108025&PAGE=Document |
00:30.04 | Gabriel25 | I read here |
00:30.09 | [TK]D-Fender | Gabriel25 : Go read up on how to add H.323 to * first. Right after confirming if you can flash taht phone to SIP. |
00:31.09 | Gabriel25 | Ok :) |
00:31.21 | Gabriel25 | let me have some readling and then I`ll be back |
00:31.51 | jeffpc | [TK]D-Fender: alright, I loaded the ztdummy module |
00:34.33 | jeffpc | I assume I need to load chan_zap.so? |
00:34.37 | [TK]D-Fender | jeffpc : good... ready to go then |
00:34.45 | Gabriel25 | [TK]D-Fender can you give me a hint ? where I can add H.323 |
00:34.55 | [TK]D-Fender | jeffpc : yup |
00:35.24 | jeffpc | [TK]D-Fender: I added load => chan_zap.so to modules.conf.. |
00:35.24 | [TK]D-Fender | Gabriel25 : Your hint.... www.voip-info.org |
00:35.28 | jeffpc | <PROTECTED> |
00:35.28 | jeffpc | Jun 5 20:33:19 WARNING[11272]: loader.c:499 load_modules: Loading module chan_zap.so failed! |
00:35.59 | [TK]D-Fender | jeffpc : wipe out your modules folder completely and redo "make install" for zaptel & asterisk |
00:36.12 | jeffpc | [TK]D-Fender: this is debian package |
00:36.13 | opus_ | jeffpc, run ztcfg -vvv |
00:36.20 | [TK]D-Fender | jeffpc : And I really hope you redid zaptel FIRST, and then recompiled asterisk afterwards... |
00:36.25 | jeffpc | duh |
00:36.29 | jeffpc | no /etc/zaptel.conf |
00:36.34 | [TK]D-Fender | jeffpc : SCrew packaging.. it'll only lead to trouble |
00:36.42 | jeffpc | [TK]D-Fender: hehe |
00:37.13 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
00:37.13 | *** mode/#asterisk [+o denon] by ChanServ |
00:37.37 | opus_ | jeffpc, there isn't a stable enough version of asterisk to package yet |
00:37.57 | opus_ | if you look at the code you will be shitfaced:) |
00:38.10 | jeffpc | opus_: I heard bad things about the code.. |
00:38.22 | jeffpc | but I'm not in the mood to compile it :) |
00:38.29 | opus_ | jeffpc: it is very bad :( |
00:38.40 | opus_ | return ""; |
00:38.45 | jeffpc | O_o |
00:38.46 | opus_ | goto outerloop: |
00:38.56 | jeffpc | o_O |
00:39.38 | [TK]D-Fender | jeffpc : Use the Source Luke! |
00:40.11 | [TK]D-Fender | jeffpc : You'll recompile your kernel, do everything else but won't do this one important easy bit? Ridiculous. |
00:40.29 | Gabriel25 | openh323 this is good ? |
00:40.35 | Gabriel25 | I have no idea sorry |
00:41.40 | jeffpc | [TK]D-Fender: :) |
00:41.48 | [TK]D-Fender | Gabriel25 : Ok, go experiment on your own a bit.. this is to Q&A for where you are at. You haven't read up on this much at all and seem to be expecting everyone else to figure it out for you fron scratch.... |
00:42.07 | opus_ | Gabriel, its crap |
00:43.06 | Gabriel25 | http://astrecipes.net/?n=102 this how to is good ? |
00:43.08 | opus_ | Gabriel, if you want a really good solid implementation you will need to hire somebody who is an expert with it |
00:43.29 | Gabriel25 | opus_ is only for me at home and just for test |
00:43.39 | Gabriel25 | It`s working fine with soft phone |
00:43.52 | [TK]D-Fender | Gabriel25 : Get off your but and just try stuff! |
00:44.05 | Gabriel25 | but know I want to add an avaya 4624 |
00:44.18 | Gabriel25 | [TK]D-Fender i like to learn stuff |
00:44.22 | Gabriel25 | And I need a start |
00:44.33 | Gabriel25 | I installed one week ago asterisk |
00:44.40 | Gabriel25 | And I made it work |
00:44.47 | Gabriel25 | with soft phone |
00:44.53 | [TK]D-Fender | Gabriel25 : 99% of * users are smart enough to avoid H.323 like the plague, you work on it for a few HOURS on your own The odds that someone here can help you at any time is pretty low. |
00:45.18 | [TK]D-Fender | Gabriel25 : Well... I guess you should have researched more before finding out now * doesn't come with H.323 by default.... |
00:45.32 | Gabriel25 | [TK]D-Fender I don`t want to use H.323 I want to use sip |
00:45.43 | [TK]D-Fender | Gabriel25 : Well go flash that phone to SIP if you can.... |
00:45.58 | Gabriel25 | this is what I`m trying to find |
00:46.14 | [TK]D-Fender | and go download the manuals, google up settings info.... just don't nag in here knowing that you have hardly tried at all.. |
00:46.14 | Gabriel25 | a firmware for that phone fo can register sip ! |
00:46.15 | Gabriel25 | :) |
00:46.34 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
00:46.42 | Gabriel25 | [TK]D-Fender you are right :) I need toearn more ! |
00:46.49 | Gabriel25 | learn more |
00:46.55 | Gabriel25 | then to ask questions |
00:47.20 | Gabriel25 | on the avaya website they don`t have |
00:47.32 | Gabriel25 | and now I have to find out if is working |
00:47.33 | Gabriel25 | :D |
00:47.48 | Gabriel25 | ftp://ftp.avaya.com/incoming/Up1cku9/tsoweb/ip_telephone/022006/4624_12_06readme_1_8_3.txt |
00:51.29 | *** join/#asterisk coppice (n=chatzill@44.199.17.210.dyn.pacific.net.hk) |
00:52.37 | *** join/#asterisk Twister (n=bob@host79.poky900.ncn.net) |
00:53.29 | Twister | hi all |
00:55.08 | Twister | im lookin for some feedback on the best softphone |
00:55.41 | [TK]D-Fender | Twister : They all suck, but eyebeam is the most functional of all. |
00:55.42 | mjh001 | idefisk |
00:55.57 | dlynes_office | [TK]D-Fender: and snom360 |
00:56.10 | Twister | from x10? |
00:56.13 | [TK]D-Fender | Twister : Why are you looking for a soft-phone at all? |
00:56.25 | *** join/#asterisk Lino` (n=Lino@i577BDDCD.versanet.de) |
00:56.38 | dlynes_office | [TK]D-Fender: causest the coolest new gadget for your desktop to infest your system tray! |
00:56.56 | Twister | my boss wants a softphone for when he is out of the office and he wants a headset that he can anwser/end calls from |
00:57.00 | [TK]D-Fender | Twister : Yes, from XTen |
00:57.24 | Twister | ok that wont work then, unfortuinatly, im looking at this headset |
00:57.32 | [TK]D-Fender | Twister : Good reason. If you don't need video and you want the perks, get X-Pro. |
00:57.41 | Twister | the plantronics CS50-USB |
00:57.57 | Twister | but dont have to get that one |
00:58.03 | [TK]D-Fender | Twister : It should work fine with that.... |
00:58.24 | Twister | the only requirment is that it be wireless and be able to anwser/end calls from the headset |
00:58.28 | opus_ | are there any good call center USB headsets that support high quality audio? |
00:59.06 | [TK]D-Fender | Twister : Oh... hrm... not sure about that part.... |
00:59.19 | Twister | ya |
00:59.22 | [TK]D-Fender | opus_ : define high quality audio.... |
00:59.46 | Twister | see that headset will do it but wont work with that softphone or so they say just a sec ill get you a link with a list |
00:59.59 | [TK]D-Fender | Twister : Ask him if he'd like fries with that.... |
00:59.59 | coppice | well, there ain't much point in it sounding better than a phone :-\ |
01:00.04 | Twister | http://www.plantronics.com/media/downloads/PerSonoCallSoftphoneCompatibility.pdf |
01:00.24 | [TK]D-Fender | coppice : I was getting there... leave a little bait first :) |
01:01.11 | [TK]D-Fender | Twister : Good.. works with eyeBeam.. there you go... |
01:01.13 | [TK]D-Fender | Twister : Problem solved. |
01:01.17 | Twister | oh |
01:01.23 | Twister | i didnt see that! whooo |
01:02.26 | mitcheloc | i got that headset |
01:02.30 | mitcheloc | it's not so good as a wired one |
01:03.15 | Twister | ya but unfortuinatly the wired one is not an option |
01:05.11 | mjh001 | Is it possable to make asterisk dial from an email, ie 1115550000-3030@asteriskbox.net would make it dial out on a voip/pstn trunk to 1115550000 and also rign extenion 3030 internaly? Has anoyone seen anything like this done befor, so fat the closet thing I've been able to find is a hosted srevice that had a web dialer where a site visitor enters a phone number and there system call them and rings another phone numbr. |
01:06.18 | Twister | mitcheloc, is there another solution you would reccomend (as far as the hardware goes) |
01:06.28 | mitcheloc | mjmac: was that hosted service asterisk compatible? |
01:06.41 | mitcheloc | Twister: not specifically, just saying wired = much better quality |
01:07.11 | mitcheloc | mjmac: does it *have* to be initiated from an e-mail? or would 3rd party software work for you? |
01:07.23 | Twister | oh i agree |
01:07.33 | *** join/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg) |
01:07.40 | *** part/#asterisk jeffpc (n=jeffpc@ool-18ba4c2d.dyn.optonline.net) |
01:08.35 | mjh001 | I do not belive so, the hosted service basicly connects a pstn to a pstn and is invoked from a web page. What I want to do is make my asterisk ring a local ext. and call out to a pstn over voip |
01:09.00 | mitcheloc | mjmac: 3rd party software okay? |
01:09.40 | mjh001 | A 3rd party app would work so ling as it can be automated and perhaps receive it direction from an email |
01:09.45 | hads | mjh001: You could just forward a local mail alias to a script and have that script generate a call file. |
01:09.54 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-2-21.red.bezeqint.net) |
01:10.37 | mitcheloc | is this webmail or outlook? |
01:12.59 | mjh001 | The remote user might be using either, or it might be from a form on a website, it would come in via smtp so the remote client dosent matter, as long as they formated it correctly. hads ->do you know of a script that will do the trick? I've been looking for about two weeks for a way to do this. |
01:13.48 | mitcheloc | mjmac: it is possible, but you'll likely need to write your own script |
01:13.58 | hads | mjh001: No sorry, I don't know of a pre-written script, it shouldn't be too hard to make though. |
01:15.31 | mjh001 | okay, thanks anyway... I guess it is time I start learning to program anyway and stop wasteing so much time searching.... Guess this will be a good place to start... |
01:15.32 | viler | Somebody that can help me in the Rhino Channel bank configuration please. |
01:16.13 | mitcheloc | mjh0001: it should be very simple, if you run an smtpd on the server or apache and php |
01:18.26 | mjh001 | okay, well I think I just pulled the wrong svn update to update my asterisk... I ended up with "Asterisk SVN-branch-1.2-r32373" off to go and look... |
01:20.32 | mjh001 | yup, I pulled a branch abd not the trunk... duh! |
01:23.31 | [TK]D-Fender | viler : Thats almost a contradiction in terms.. Rhino's autodetect jsut about anything you through at them |
01:24.31 | *** join/#asterisk denon (n=denon@synapse.subneural.net) |
01:24.31 | *** mode/#asterisk [+o denon] by ChanServ |
01:25.01 | opus_ | Jun 5 18:24:24 WARNING[5637]: chan_sip.c:2542 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4) |
01:25.06 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.207) |
01:25.13 | opus_ | anyone seen that one before? |
01:26.13 | dasenjo | Hi, I'm trying to compile * 1.0.11 and got this error: http://pastebin.ca/62367 |
01:26.20 | dasenjo | can you help me? |
01:28.19 | russellb | dasenjo: i don't see anything wrong with that gcc command |
01:28.33 | russellb | and 1.0.11 builds fine for me. |
01:28.58 | dasenjo | russellb, this is a "dirty" system .. |
01:29.04 | russellb | i can tell :-p |
01:29.05 | [TK]D-Fender | russellb : Well it does say : gcc: cannot specify -o with -c or -S and multiple compilations |
01:29.18 | [TK]D-Fender | russellb : Bad GCC version maybe? |
01:29.24 | russellb | yeah |
01:29.31 | dasenjo | I have installed *-1.2 from packages .. a |
01:29.36 | russellb | or ... something |
01:29.43 | dasenjo | and another * zap and libpri form sources . |
01:29.47 | russellb | well if you are running 1.2, why are you trying to install 1.0.11? |
01:29.49 | file | so why are you downgrading to 1.0.11? |
01:30.04 | russellb | file: jinx. |
01:30.07 | dasenjo | fxo interaction problems .. |
01:30.20 | dasenjo | i could not make work 1.2 .. |
01:30.21 | russellb | downgrading will not likely help you |
01:30.25 | russellb | is it a digium card? |
01:30.35 | dasenjo | 1.0.10 works very fine .. |
01:30.40 | dasenjo | TDM400P |
01:30.46 | russellb | have you contacted digium support? |
01:31.00 | dasenjo | no |
01:31.04 | russellb | well why not? |
01:31.23 | dasenjo | I live in ther "third world" |
01:31.44 | russellb | i see ... well it looks like your system needs some upgrading :) |
01:32.40 | dasenjo | it is a debian stable without problems .. |
01:32.52 | russellb | well you said dirty |
01:32.57 | russellb | i figured it was something terribly out of date |
01:33.30 | [TK]D-Fender | dasenjo : Forget 1.0.X and jsut install the latest release from FTP (1.2.9) |
01:33.47 | dasenjo | no .. so many installations form source .. that's all .. |
01:33.57 | dasenjo | [TK]D-Fender, it does not work .. |
01:34.03 | dasenjo | for my conf. |
01:34.26 | dasenjo | or for my card/line combination .. |
01:34.50 | dasenjo | just 1.0.10 with MARK2 and AGRESSIVE_SUPPRESSION works to detect hang up .. |
01:37.04 | dasenjo | this is a very strange error .. asterisk 1.0.11 _should_ compile |
01:39.38 | opus_ | decodeMP3: Junk at the beginning of frame 00000000 |
01:39.45 | opus_ | anyone know of a solution around that problem? |
01:39.56 | *** join/#asterisk NewSole (n=dave@d226-107-112.home.cgocable.net) |
01:40.35 | NewSole | Wee... After Months of Work.... we now Have a Working softphone... weeee..... |
01:45.42 | *** join/#asterisk ceeto (i=cio@adsl-072-149-159-016.sip.bhm.bellsouth.net) |
01:45.54 | ceeto | Hi all. I have like 15 asterisk processes running, is that normal? |
01:46.11 | b00mer | I am trying to setup a working screen, someone earlier said I should check in to the privacy settings in app_dial... anyone have a working example? |
01:46.11 | [TK]D-Fender | ceeto : not processes, threads |
01:46.30 | [TK]D-Fender | ceeto : I know it looks funy.. you see it more on 2.4 kernels I believe.... |
01:46.40 | [TK]D-Fender | ceeto : but everything is fine |
01:46.55 | ceeto | Does each process take memory? |
01:47.08 | b00mer | but the privacymanager doesn't look like what I am looking for |
01:47.16 | [TK]D-Fender | ceeto : Clearly... how much I couldn't say |
01:47.17 | russellb | all of the threads share the same memory |
01:47.39 | russellb | it's not the same as multiple processes ... |
01:47.51 | b00mer | anyone available to help debug / understand my macro issue? |
01:48.15 | [TK]D-Fender | b00mer : pastebin away... everything related to it. |
01:48.21 | b00mer | http://pastebin.com/760744 |
01:48.33 | b00mer | What I am trying to accomplish is: |
01:48.38 | *** join/#asterisk Hmmhesays (n=Neg@31-201.69-92-cpe.cableone.net) |
01:48.58 | Hmmhesays | this embedded sh1at sucks |
01:49.02 | b00mer | Someone calls a support number... 4 engineers are called with a short message saying press 1 if you want to take the calll |
01:49.08 | b00mer | right now it works, but |
01:49.20 | b00mer | if they pickup and let it time out... fine |
01:49.31 | b00mer | if they pick up and hangup it hangs up on the caller |
01:49.42 | b00mer | if they pickup and press 1 they are connected... fine |
01:49.56 | b00mer | so I am only having issue with the pickup and hangup |
01:50.18 | Hmmhesays | using a queue? |
01:50.22 | b00mer | no queues |
01:50.36 | b00mer | just using this "screeen" logic |
01:50.59 | [TK]D-Fender | b00mer : What would you have it do on pickup/hangup? |
01:52.43 | b00mer | the CONTINUE |
01:52.52 | *** join/#asterisk asterisk-dud (n=dwwollma@64-42-247-120.mb.skyweb.ca) |
01:53.09 | b00mer | unless they press 1 ... I want it to assume that the call is not wanted and continue on the call list |
01:53.27 | asterisk-dud | i'm using a tdm405p t1 card and i get a whistling sound for a dialtone, |
01:53.49 | asterisk-dud | anyone know what could be wrong? |
01:53.54 | matthewsimpson | you bought the bluebox edition tdm405p :( |
01:53.58 | *** join/#asterisk TESTER2 (n=Cyber@modemcable082.42-81-70.mc.videotron.ca) |
01:54.15 | asterisk-dud | what is bluebox edition? |
01:54.20 | russellb | there is no such thing as a tdm405p :) |
01:54.28 | TESTER2 | any idea about a very old phone (inband alimentation) not riging on a spa-1001? |
01:54.58 | asterisk-dud | te405p |
01:54.59 | asterisk-dud | sorry |
01:54.59 | [TK]D-Fender | b00mer : add an "h" exten or force the dial to continue on (can't recall the parm for that) |
01:55.43 | asterisk-dud | russellb it's te405p |
01:56.03 | russellb | asterisk-dud: gotcha. it's most likely a problem with zone settings in /etc/zaptel.conf |
01:56.35 | surfdue | bk |
01:56.44 | asterisk-dud | ok, i was wondering about that, currently i have nothing for that, but it worked like that before |
01:56.47 | dlynes_office | [TK]D-Fender: I found out why sangoma's not getting compiled |
01:56.54 | b00mer | [TK]D-Fender: you think a exten => h,1,SetVar(MACRO_RESULT=CONTINUE) will work? |
01:56.59 | [TK]D-Fender | dlynes_office : oh, do tell :) |
01:57.02 | dlynes_office | [TK]D-Fender: its dependencies aren't getting satisfied in the kernel build |
01:57.12 | *** join/#asterisk jeffik (n=Jeff@Maroon-103-176.ADSL.NetSurf.Net) |
01:57.14 | dlynes_office | [TK]D-Fender: and so i never have the option to select sangoma in the make menuconfig |
01:57.14 | russellb | asterisk-dud: well, you need to add those options :) |
01:57.21 | [TK]D-Fender | b00mer : Give it a shot. I'm not sure if that will get called in the macro, or in the context with your Dial in it... |
01:57.50 | asterisk-dud | would that be y the first hole doesn't go away from a red alarm? |
01:57.52 | [TK]D-Fender | brb, gotta restart my server.. X went haywaire... |
01:58.27 | russellb | asterisk-dud: um, probably not ... |
01:58.45 | asterisk-dud | how would i know if my card is dunked |
01:59.01 | russellb | contact tech support, but it's probably fine ... |
01:59.31 | asterisk-dud | should i add the location line at the end or at the beginning or isn't it a big deal |
01:59.55 | russellb | i would put it before the channels, because i can't remember if it matters |
02:02.07 | asterisk-dud | russellb it didn't help |
02:02.25 | russellb | i don't believe you :) |
02:02.28 | Hmmhesays | just what I needed (just what I needed) |
02:02.37 | asterisk-dud | that's terrible |
02:02.38 | russellb | did you reconfigure the card using ztcfg ? |
02:02.38 | asterisk-dud | lol |
02:02.42 | asterisk-dud | yes |
02:03.00 | Hmmhesays | did you swear at it? |
02:03.20 | *** part/#asterisk TESTER2 (n=Cyber@modemcable082.42-81-70.mc.videotron.ca) |
02:03.20 | asterisk-dud | loadzone= us |
02:03.21 | asterisk-dud | defaultzone= us |
02:05.18 | *** join/#asterisk kernel20 (n=kernel20@203.160.223.26) |
02:05.24 | [TK]D-Fender | . |
02:05.30 | [TK]D-Fender | Ok, not cut off it seems |
02:05.32 | kernel20 | hi what is MACRO_EXTEN? |
02:05.32 | Hmmhesays | anyone ever used buildroot for ulibc? |
02:05.36 | Hmmhesays | *uclibc |
02:05.56 | [TK]D-Fender | Wow and it jsut caught me up on everything I missed :) |
02:06.16 | Hmmhesays | what are you babbling about [TK]D-Fender? |
02:06.23 | kernel20 | most of the manuals on voicemails deals on MACRO_EXTEN |
02:06.23 | russellb | asterisk-dud: please contact digium technical support, they will be able to help you |
02:07.10 | kernel20 | exten => s-NOANSWER,1,Voicemail(u${MACRO_EXTEN} will anounce user is unavailable, now the problem is the MACRO_EXTEN, how can i relate it in my voicemail.conf? |
02:07.30 | [TK]D-Fender | kernel20 : Go read up on macro's..... |
02:07.41 | dlynes_office | [TK]D-Fender: i only told it to compile tdm drivers last time...let's go see what it says when i tell it all drivers |
02:07.49 | dlynes_office | [TK]D-Fender: try this one out |
02:07.52 | dlynes_office | ~kernel20 |
02:07.56 | jbot | i guess kernel20 is an annoying user that is allergic to reading documentation. |
02:07.56 | kernel20 | i am reading, but got troubles on understanding MACRO_EXTEN |
02:07.56 | [TK]D-Fender | dlynes_home : um.. no |
02:08.15 | kernel20 | please dont start |
02:08.41 | kernel20 | anybody else? |
02:09.32 | dlynes_office | [TK]D-Fender: ? |
02:09.37 | [TK]D-Fender | kernel20 : this one is a freebie : When you call a macro like "exten => 110,1,Macro(justamacro,1234)" ${MACRO_EXTEN} = 110 and ${ARG1} = 1234. Got it? |
02:09.37 | kernel20 | where should i declare macro_exten? and how can i relate it to my voicemail.conf |
02:09.53 | asterisk-dud | anyone using channel banks? |
02:10.07 | [TK]D-Fender | kernel20 : You don't declare it. its set by the line that calls the macro. It is a concept you are not required to implement. |
02:10.20 | [TK]D-Fender | asterisk-dud : I use a Rhino modular CB. |
02:11.01 | asterisk-dud | have u ever used adtran |
02:11.07 | littleball | hello, i am looking for SER/asterisk reference deployment notes. who can help? |
02:11.20 | [TK]D-Fender | asterisk-dud : nope. |
02:11.36 | asterisk-dud | r u fxs or fxo D-Fender |
02:11.56 | kernel20 | [TK]D-Fender: how can i set it |
02:11.57 | kernel20 | ? |
02:13.09 | *** join/#asterisk IBN287-Jo (i=anpu359@pool-72-65-231-46.pitbpa.east.verizon.net) |
02:13.25 | *** part/#asterisk IBN287-Jo (i=anpu359@pool-72-65-231-46.pitbpa.east.verizon.net) |
02:13.36 | [TK]D-Fender | kernel20 : You don't set it. It gets set by the line that calls the macro. Look at my sample and open your eyes. |
02:13.46 | [TK]D-Fender | asterisk-dud : I run both |
02:14.13 | kernel20 | [TK]D-Fender: http://pastebin.com/761306 <- how can i add voicemail on it? |
02:14.23 | asterisk-dud | ok, for fxs ports, i use fxo_ks signalling correct? |
02:15.21 | mitcheloc | kernel20 came back? |
02:15.32 | kernel20 | [TK]D-Fender: what did u see? |
02:15.34 | [TK]D-Fender | kernel20 : use your imagination, and when those 5 seconds are up, go read this and pray for inspiration : http://www.voip-info.org/wiki/view/Stdexten+macro |
02:15.44 | mitcheloc | ~kernel20 |
02:15.45 | jbot | from memory, kernel20 is an annoying user that is allergic to reading documentation. |
02:15.49 | [TK]D-Fender | asterisk-dud : Correct |
02:16.21 | [TK]D-Fender | mitcheloc : Oh just put a lid on it. You aren't part of the solution... |
02:16.34 | kernel20 | http://pastebin.com/761306 <- how can i add voicemail on it? base on that link |
02:16.48 | *** join/#asterisk `Kevin (n=Kevin@64.243.236.20) |
02:16.51 | [TK]D-Fender | kernel20 : Did you go to the WIKI page I linked? |
02:17.12 | mitcheloc | [TK]D-Fender: =P, i'd help, but i've tried before |
02:17.24 | kernel20 | yeah |
02:17.25 | mitcheloc | [TK]D-Fender: i believe kernel20 called me an "ass" when i gave him a suggestion |
02:17.38 | kernel20 | but really dont get the whole point sorry |
02:17.45 | kernel20 | mitcheloc: please dont start |
02:17.46 | [TK]D-Fender | kernel20 : Go see how they did VM for it, and just do the same. |
02:18.01 | [TK]D-Fender | mitcheloc : Thats nice. Now put a lid on it. |
02:18.10 | kernel20 | [TK]D-Fender: http://pastebin.com/761306 <- how can i add voicemail on it? base on that link |
02:18.24 | kernel20 | please look at the link |
02:18.33 | kernel20 | the voicemail is at voicemail.conf |
02:18.38 | asterisk-dud | D-Fender, if my channel bank is FXS Loop Start, should i use FXS_ks signalling |
02:18.59 | kernel20 | my problem is how would it able to access the declarations i have done at voicemail.conf |
02:19.05 | [TK]D-Fender | kernel20 : When do you want to go to VM in there? |
02:19.18 | [TK]D-Fender | asterisk-dud : fxo_ks |
02:19.24 | kernel20 | if NOANSWER |
02:19.35 | [TK]D-Fender | kernel20 : then add that. |
02:19.50 | kernel20 | yeah but it wont prompts for voicemail |
02:20.00 | [TK]D-Fender | kernel20: DO IT then show me. |
02:20.02 | kernel20 | please have a look |
02:20.33 | [TK]D-Fender | kernel20 : I did have a look . CHANGE IT YOURSELF RIGHT NOW. try something and then see what doesn't work. |
02:20.46 | kernel20 | http://pastebin.com/761315 |
02:21.00 | kernel20 | [TK]D-Fender: http://pastebin.com/761315 |
02:21.21 | [TK]D-Fender | kernel20 : Good start... now tell me... what value do you imagine is in ${ARG1} ? |
02:21.38 | kernel20 | SIP/11000 |
02:21.49 | kernel20 | i want to direct it to voice mail |
02:21.49 | [TK]D-Fender | kernel20 : is that a valide mailbox name? |
02:21.55 | kernel20 | no |
02:21.59 | kernel20 | thats my problem |
02:22.09 | [TK]D-Fender | kernel20 : then you need to apss it ANOTHER argument that holds a useful value |
02:22.10 | kernel20 | base on that how can i relate it to my voicemail |
02:22.22 | [TK]D-Fender | ${ARG2} |
02:22.26 | kernel20 | can u add sample on that link? |
02:22.35 | [TK]D-Fender | kernel20 : pass 2 parameters in your macro. |
02:22.49 | dlynes_office | [TK]D-Fender: got something loading now :) |
02:22.50 | kernel20 | ahhh |
02:22.54 | kernel20 | hehehe |
02:22.56 | kernel20 | got it now |
02:22.57 | [TK]D-Fender | kernel20 : No. I want you to listen to my rather direct advice and do it yourslef, then show me. |
02:22.58 | kernel20 | too dumb |
02:22.59 | kernel20 | hahahah |
02:23.05 | kernel20 | wait |
02:23.39 | *** join/#asterisk pigpen2 (n=mark@fw.seamans.cc) |
02:24.18 | dlynes_office | [TK]D-Fender: for whatever reason, the sangoma stuff isn't showing up in make menuconfig |
02:24.56 | surfdue | my software cant connect to the server im not behind a nat nor is my server |
02:25.00 | surfdue | why is this? |
02:25.09 | surfdue | can I use a website to test the connection to asterisk? |
02:25.10 | dlynes_office | surfdue: errors? |
02:25.15 | surfdue | maybe |
02:25.18 | surfdue | where are the logs again |
02:25.19 | surfdue | i forget |
02:25.21 | [TK]D-Fender | dlynes_home : What is this menuconfig of which you speak? |
02:25.22 | dlynes_office | surfdue: /var/log/asterisk |
02:25.24 | surfdue | /var/log |
02:25.25 | surfdue | ok |
02:25.28 | dlynes_office | [TK]D-Fender: for the linux kernel |
02:25.47 | dlynes_office | [TK]D-Fender: you know? make menuconfig/make config/make xconfig |
02:25.49 | [TK]D-Fender | dlynes_home : Why are you screwing with your kernel? |
02:26.15 | dlynes_office | [TK]D-Fender: to enable the options that are necessary for the sangoma |
02:26.39 | *** join/#asterisk rvhi (n=rv@66.175.65.89) |
02:27.09 | [TK]D-Fender | dlynes_home : Ummm.. I never ran into an out of the box setup that needed anything.... |
02:27.28 | kernel20 | [TK]D-Fender: http://pastebin.com/761334 |
02:27.33 | surfdue | dlynes_office, http://host41.com/log.txt |
02:27.38 | kernel20 | please do have a check |
02:27.45 | dlynes_office | [TK]D-Fender: yeah, but if change something in my kernel, i don't want to have to go through the whole sangoma setup again |
02:27.46 | kernel20 | it doesnt work what i expected |
02:27.48 | surfdue | would those cause it to not beable to connect? |
02:27.58 | NewSole | Wee... We now Have a Working softphone... weeee..... |
02:28.22 | [TK]D-Fender | kernel20 : You reloaded this new config and simply didn't answer the call? |
02:28.24 | kernel20 | [TK]D-Fender:? |
02:28.37 | kernel20 | i reloaded |
02:28.45 | kernel20 | it that conf workable?. |
02:28.47 | dlynes_office | surfdue: ummmmare you using the sample config file? |
02:28.56 | surfdue | ya |
02:29.00 | surfdue | well no |
02:29.09 | dlynes_office | surfdue: which is it? yes, or no? |
02:29.11 | surfdue | the default ones in /etc/asterisk |
02:29.17 | surfdue | after make samples |
02:29.20 | dlynes_office | yeah...so you're using hte sample config files |
02:29.21 | [TK]D-Fender | kernel20 : pastebin a call attempt |
02:29.46 | dlynes_office | surfdue: lose all the extra crud from sip.conf and extensions.conf that you're not using |
02:29.54 | surfdue | really |
02:29.55 | surfdue | ? |
02:29.59 | surfdue | like even the bind stuff |
02:30.00 | surfdue | and all that? |
02:30.09 | dlynes_office | surfdue: use your better judgement |
02:30.19 | surfdue | yes? |
02:30.27 | dlynes_office | surfdue: you can keep the stuff in general |
02:30.34 | dlynes_office | surfdue: but anything else you're not using, get rid of it |
02:30.57 | dlynes_office | surfdue: or if you want, pastebin it, and pastebin it back with all the crap taken out |
02:31.18 | dlynes_office | surfdue: you'll find it very difficult to solve your problems in asterisk when your config files are a mess |
02:31.31 | dlynes_office | surfdue: especially when they have a bunch of crap in them that you don't understand |
02:31.36 | surfdue | done |
02:32.00 | surfdue | dlynes_ will you do it? |
02:32.03 | surfdue | if i pastebin it |
02:32.09 | dlynes_office | surfdue: yes...just pastebin it |
02:32.16 | surfdue | with the password? |
02:32.21 | dlynes_office | surfdue: are you using asterlink as your voip provider? |
02:32.26 | dlynes_office | surfdue: no...scrub the passwords |
02:32.35 | surfdue | ya |
02:32.47 | kernel20 | [TK]D-Fender: http://pastebin.com/761343 |
02:32.48 | dlynes_office | surfdue: i.e. replace the passwords with XXXXX or something similar |
02:33.41 | [TK]D-Fender | kernel20 : Look at your Goto. theres the problem. |
02:33.54 | *** join/#asterisk xachen (i=justin@pdpc/supporter/student/xachen) |
02:34.04 | surfdue | http://host41.com/sip.conf |
02:34.32 | kernel20 | [TK]D-Fender: what is problem? |
02:34.47 | asterisk-dud | when i start up my channel bank all 24 lines flash and my phones start ringing |
02:35.01 | dlynes_office | cool trick, asterisk-dud |
02:35.07 | [TK]D-Fender | kernel20 : Look at your Goto VERY closely and then again at your dial-lan and you'd beet er see for yourself whats wrong.... |
02:35.24 | kernel20 | what is the problem |
02:35.28 | kernel20 | i am very close now |
02:35.33 | kernel20 | please tell me |
02:35.46 | [TK]D-Fender | kernel20 : try using your eyes... where is the Goto "going"? |
02:35.48 | asterisk-dud | dlynes-office it's pissing me off |
02:35.58 | asterisk-dud | i don't think i could if i tried |
02:36.19 | kernel20 | to my next line |
02:36.21 | kernel20 | of course |
02:36.40 | dlynes_office | asterisk-dud: i would imagine your span is incorrect or something |
02:36.47 | dlynes_office | asterisk-dud: are you using a four port pri card? |
02:37.27 | *** join/#asterisk tsurk0 (n=tsurko@85.187.160.157) |
02:37.32 | [TK]D-Fender | kernel20 : Read this and WAKE UP!!!! -- Executing Goto("SIP/11000-a4a9", "s-NOANSWER|1") in new stack |
02:37.37 | dlynes_office | surfdue: http://pastebin.com/761345 |
02:37.46 | asterisk-dud | a te405p |
02:38.07 | dlynes_office | asterisk-dud: you've got the channel bank plugged into the top port, or the bottom port? |
02:38.23 | asterisk-dud | top one, |
02:38.35 | dlynes_office | asterisk-dud: pastebin your zaptel.conf file |
02:38.51 | asterisk-dud | where do i pastebin? |
02:38.56 | asterisk-dud | i'm new at this |
02:39.01 | dlynes_office | ~pb |
02:39.03 | jbot | pb is probably a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
02:39.29 | kernel20 | ? |
02:39.44 | kernel20 | [TK]D-Fender: please pastebin |
02:40.01 | [TK]D-Fender | kernel20 : Wake up! there is no "s-NOANSWER|1" in your macro for it to Goto to. |
02:40.27 | dlynes_office | [TK]D-Fender: a prime reason why peeps shouldn't use auto fallthrough :) |
02:40.42 | [TK]D-Fender | dlynes_office : Irrelevent to debugging this,. |
02:40.50 | surfdue | ty dlynes |
02:40.53 | asterisk-dud | dlynes_office: http://pastebin.com/761346 |
02:40.57 | [TK]D-Fender | dlynes_office : the problem is BLATANTLY obvious |
02:41.08 | dlynes_office | [TK]D-Fender: yeah, but it would help kernel20 understand where his problems lie, easier |
02:41.12 | dlynes_office | [TK]D-Fender: yeah, i realize that |
02:41.16 | topping | anyone have favorite IAX-PSTN providers for the US? |
02:41.34 | dlynes_office | [TK]D-Fender: that's why i usually have him on /ignore |
02:41.37 | [TK]D-Fender | dlynes_office : you mean those list 2 lines I made weren't obvious enough? *sigh* |
02:42.13 | asterisk-dud | dlynes_office: did u get that? |
02:42.19 | dlynes_office | asterisk-dud: yeah |
02:42.30 | NewSole | We need some Beta Testers..... Any takers |
02:42.48 | surfdue | MySQL RealTime: Failed to connect database server on . |
02:42.51 | surfdue | how do i fix this? |
02:43.25 | dlynes_office | asterisk-dud: hrm...i'm not sure how to fix your problem...the loadzone and defaultzone and span should all be fine for a channel bank |
02:43.46 | dlynes_office | asterisk-dud: but i'm not sure about the fxoks; i'm not sure if it should be that, or channels specified |
02:43.47 | surfdue | i think its missing a mysql conf file |
02:43.53 | surfdue | Jun 5 22:41:24 WARNING[12374] cdr_addon_mysql.c: Unable to load config for mysql CDR's: cdr_mysql.conf |
02:43.55 | surfdue | where do i get this? |
02:43.56 | dlynes_office | asterisk-dud: i've never configured a channel bank |
02:44.04 | asterisk-dud | could there be a prob with my zapata? |
02:44.10 | dlynes_office | surfdue: go into modules.conf and do an unload => cdr_mysql.so |
02:44.20 | surfdue | but i want mysql? |
02:44.34 | dlynes_office | surfdue: worry about it later after you get everything working |
02:44.44 | [TK]D-Fender | asterisk-dud : pastebin your zapata. And your zaptel should be 1,1,0 not 1,0,0 |
02:44.47 | dlynes_office | surfdue: you'll need to configure the database, set it up, get it running and everything else |
02:45.06 | surfdue | how? |
02:45.09 | surfdue | i wanna do this now |
02:45.09 | surfdue | :P |
02:45.12 | surfdue | i need help : |
02:45.13 | surfdue | :P |
02:45.28 | dlynes_office | surfdue: you're talking to someone that's never set it up before |
02:45.37 | dlynes_office | surfdue: cdr_mysql is in asterisk-addons-1.2.3 |
02:46.04 | surfdue | which i did install |
02:46.14 | dlynes_office | surfdue: well, then |
02:46.50 | dlynes_office | surfdue: you need to read up on the documentation for it, so you know what username, password to set up for the database, how to configure where the database is, what tables to create, and how to define them, ... |
02:47.03 | surfdue | i see it |
02:47.13 | dlynes_office | surfdue: it should all be in asterisk-addons |
02:47.59 | asterisk-dud | dlynes_office: i fixed |
02:48.07 | asterisk-dud | moved my card to different slot |
02:51.25 | NewSole | We need some Beta Testers..... Test New SoftPhone this week Any takers |
02:51.43 | mitcheloc | NewSole: what's it called? |
02:51.46 | dlynes_office | [TK]D-Fender: looks like i might need a firmware upgrade to get this working |
02:51.59 | NewSole | SoftPhone..... |
02:52.00 | [TK]D-Fender | dlynes_office : Good idea as well.... |
02:52.20 | mitcheloc | heh, neutral name ;) |
02:52.20 | [TK]D-Fender | dlynes_office : Make sure to get the EC tools as well if you have that module. |
02:52.23 | mitcheloc | i'll help test if you want |
02:52.39 | dlynes_office | [TK]D-Fender: yeah...i've got that for the a200d with the ec |
02:52.55 | dlynes_office | [TK]D-Fender: it's probing fine...just can't allocate PIC memory on the card |
02:52.55 | NewSole | its supports SIP/H323/IAX2 and has built in G729/G723/G726/ULAW/ALAW/GSM/iLBC/SPEEX |
02:53.15 | mitcheloc | why would you add h323? |
02:53.26 | dlynes_office | mitcheloc: for connecting to Quintums? |
02:53.31 | surfdue | were are the modules located for asterisk? |
02:53.35 | mitcheloc | whats that? |
02:53.40 | dlynes_office | mitcheloc: and for certain providers that only do h323? |
02:53.47 | mitcheloc | who? |
02:53.59 | dlynes_office | mitcheloc: Sun Telecom is one I can think of off the top of my head |
02:54.03 | NewSole | also has built in call Forwaring and Voice Mail Recording |
02:54.06 | dlynes_office | mitcheloc: they do h323 in Canada, SIP in the US |
02:54.23 | dlynes_office | NewSole: Windows or Linux? |
02:54.24 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
02:54.26 | mitcheloc | i think he is ignoring me |
02:54.26 | mitcheloc | heh |
02:54.30 | NewSole | Windows |
02:54.42 | dlynes_office | NewSole: Yeah, I know someone that would probably test it then |
02:54.45 | NewSole | no not ignoring u.... PM's |
02:55.17 | mitcheloc | NewSole: does it have a command line interface for dialing? |
02:55.28 | NewSole | its going to be freeware wile in beta..... but it will be 20$ when done |
02:55.38 | topping | NewSole: I would test on OS-X :-) |
02:56.05 | NewSole | its not built on Mac yet... |
02:56.16 | dlynes_office | NewSole: Yeah...my friend's definitely interested |
02:56.30 | surfdue | anyone? |
02:56.31 | dlynes_office | NewSole: Should I msg you his email address? |
02:56.41 | NewSole | Requires Microsoft .Net Framework..... (100% .Net Framework) |
02:56.41 | dlynes_office | surfdue: /usr/lib/asterisk/modules (default) |
02:56.53 | dlynes_office | NewSole: Not an issue...he's a .NET developer |
02:57.03 | mitcheloc | NewSole: command line? |
02:57.09 | NewSole | no not yet |
02:57.32 | mitcheloc | i can add support for it in Snap when you are ready with the command line =) |
02:57.59 | NewSole | it also has a built in emailer... to email the recorded voice mails |
02:58.03 | dlynes_office | NewSole: msg you email details, or is there a download page for it? |
02:58.39 | mitcheloc | lol, i feel so ignored talking to newsole =/ |
02:59.03 | NewSole | send emails requests to mworkman@imbroadcasting.net..... Beta's emailed out on Weekend |
02:59.09 | dlynes_office | NewSole: ok, thanks |
02:59.29 | NewSole | mitcheloc... I am not ignoring you |
03:00.21 | NewSole | just alot of PM's from Beta testers.... and new build comming out this weekend |
03:00.59 | dlynes_office | NewSole: it is compatible with asterisk, right? |
03:00.59 | mitcheloc | dlynes_office: tell your friend about snap, i just put out my mozilla plugins ;) |
03:01.11 | dlynes_office | mitcheloc: you mean mmc? |
03:01.12 | NewSole | not doing comand line for it.... but it will support VB Scripts when release mode done |
03:01.19 | surfdue | http://host41.com/log.txt |
03:01.21 | mitcheloc | mmc? |
03:01.24 | surfdue | can someone help me fix those errors |
03:01.28 | dlynes_office | Microsoft Management Console |
03:01.36 | mitcheloc | NewSole: why no command line? |
03:01.44 | mitcheloc | dlynes_home: no i mean, mozilla plugins |
03:01.47 | dlynes_office | mitcheloc: because it's a windows plugin? |
03:01.53 | dlynes_office | erm application i mean? |
03:02.03 | dlynes_office | mitcheloc: so what's snap then? |
03:02.09 | mitcheloc | mozilla = firefox/thunderbird |
03:02.15 | dlynes_office | is it like snap crackle pop rice crispies? |
03:02.19 | mitcheloc | it's a dialer, heh check out www.snapanumber.com |
03:02.33 | mitcheloc | i'm adding support for softphones and stuff, so they can focus on being phones and i can focus on dialing |
03:03.02 | dlynes_office | ah, cool |
03:03.21 | dlynes_office | Now if only I could convince him to get rid of Internet Exploder :) |
03:03.45 | dlynes_office | He's finally starting to get into the whole linux groove thing :) |
03:03.51 | mitcheloc | working on IE also, just not done yet |
03:03.53 | dlynes_office | He does Python .NET development |
03:03.59 | mitcheloc | wierd |
03:04.08 | dlynes_office | He loves python and loves .NET |
03:04.16 | dlynes_office | but he's not a big fan of Microsoft :0 |
03:04.23 | mitcheloc | lol, nice |
03:05.22 | asterisk-dud | i would like to builg a timer that could be overridden with a passowrd |
03:05.41 | mitcheloc | timer? |
03:06.12 | asterisk-dud | that would hang up the line |
03:06.17 | dlynes_office | mitcheloc: so they're thunderbird/firefox extensions then, not plugins |
03:06.20 | dlynes_office | mitcheloc: even better |
03:06.39 | mitcheloc | yea, i keep forgetting the difference in the terminology |
03:06.48 | dlynes_office | asterisk-dud: you would probably need to implement that in agi |
03:06.49 | mitcheloc | i'll update that |
03:07.03 | dlynes_office | mitcheloc: firefox does plugins, too....plugins are different |
03:07.19 | dlynes_office | mitcheloc: plugins are separate programs that run as a child window of firefox |
03:07.25 | asterisk-dud | i've tried with java-agi, but i think i'll have to use a different language |
03:07.28 | dlynes_office | mitcheloc: extensions are XUL applications |
03:08.06 | dlynes_office | mitcheloc: but yeah...that's a wicked extensions |
03:08.15 | dlynes_office | mitcheloc: we were looking for something like that about 2 years ago |
03:08.33 | *** join/#asterisk trixter (n=trixter@65-165-167-217.du.volcano.net) |
03:08.35 | mitcheloc | heh, well it's an xul extension then, thanks =) |
03:08.36 | dlynes_office | mitcheloc: how much are you selling it for? |
03:08.54 | dlynes_office | mitcheloc: or you're not the author? |
03:08.54 | trixter | has anyone noticed a threading issue with 1.2.9 that if you do asterisk -rx 'show channels' it displays the header but nothing else? |
03:08.55 | mitcheloc | well they are free, but i'd appreciate if you buy the pro so i keep developing =) |
03:09.00 | dlynes_office | trixter: 1.2.9? |
03:09.01 | mitcheloc | it's $29.99, yes i'm the author |
03:09.18 | trixter | yes 1.2.9 was released today to fix a security problem in chan_iax2 that lets people crash your server |
03:09.21 | trixter | 1.0.11 as well |
03:09.27 | trixter | its in the topic |
03:09.36 | dlynes_office | trixter: ah...cooll..didn't see it in my email |
03:09.41 | dlynes_office | and i never read the topic :) |
03:10.26 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
03:10.37 | dlynes_office | oh yeah...the announcements show up in my home email |
03:10.40 | mitcheloc | dlynes_office: try it out for a bit, and let me know how it goes for you ;) |
03:10.42 | trixter | I just noticed that show channels, sip show channels and iax2 show channels all fails if done from asterisk -rx with > 50 channels in use (it doesnt appear to happen at all with fewer channels) and this worked on a previous 1.2.x version so something in the last few versions broke that somehow |
03:10.46 | trixter | prolly a threading mutex issue |
03:10.58 | dlynes_office | mitcheloc: does it work in linux and windows? |
03:11.13 | mitcheloc | dlynes_home: not on linux...yet, will get to that soon |
03:11.32 | dlynes_office | mitcheloc: ah...shouldn't require much extra code if any for linux though, right? |
03:11.59 | mitcheloc | nah, just some time debugging the problem areas in my code |
03:12.14 | dlynes_office | trixter: hrm...i think i remember someone babbling about it in asterisk-dev last night |
03:12.15 | surfdue | dlynes_ can you help me clean this up http://www.host41.com/extensions.conf |
03:12.23 | surfdue | i only added 2 lines 2 of the extension lines |
03:13.33 | NewSole | so who wants to be a beta tester for next week.... email me |
03:13.58 | surfdue | NewSole, i have a server and wouldnt mind |
03:14.01 | surfdue | beta test what? |
03:14.17 | NewSole | a new windows Softphone |
03:14.38 | dlynes_office | surfdue: add newsole's new softphone and mitcheloc's dialer app together for a cool combo :) |
03:15.04 | mitcheloc | dlynes_office: yea haha, use the "Path Connection" in my dialer, you don't even need to configure it for your asterisk server ;) |
03:15.08 | NewSole | IAX/SIP/H323 with G729/G723/G726/ULAW/ALAW/GSM/iLBC/SPEEX |
03:15.13 | mitcheloc | autoconfigs for skype and vonage and eyebeam users too |
03:15.19 | surfdue | NewSole, sure |
03:15.28 | surfdue | dlynes are you able to clean that fo rme? |
03:15.34 | dlynes_office | surfdue: are you using dundi? |
03:15.37 | NewSole | send emails requests to mworkman@imbroadcasting.net..... Beta's emailed out on Weekend |
03:15.50 | surfdue | dundi? |
03:15.59 | dlynes_office | surfdue: e164 |
03:16.00 | asterisk-dud | i added a second channel bank to my t1 card and i don't get a dialtone |
03:16.07 | asterisk-dud | both lights are green |
03:16.26 | surfdue | done |
03:16.31 | surfdue | e14? |
03:16.32 | dlynes_office | asterisk-dud: if it's green then there's probably something screwy in your zapata.conf file |
03:16.34 | surfdue | whats that |
03:16.34 | surfdue | lol |
03:16.43 | surfdue | what page in the book is it? |
03:16.51 | dlynes_office | surfdue: forget it then...i'll comment it out and move it all down to the bottom of the script |
03:17.28 | asterisk-dud | can u check it? |
03:17.49 | dlynes_office | i could if you pastebinned it |
03:17.52 | surfdue | :P |
03:18.08 | surfdue | dlynes http://www.host41.com/extensions.conf |
03:18.22 | dlynes_office | surfdue: i'm already working on that |
03:18.28 | asterisk-dud | dlynes_office: http://pastebin.com/761418 |
03:19.11 | surfdue | k |
03:19.53 | dlynes_office | surfdue: you've only got asterlink for all outbound calls? |
03:20.40 | surfdue | ya |
03:20.52 | surfdue | and inbound i just wnana set it up so im on ext 200 |
03:20.59 | surfdue | and i can setup a voice menu with hold music and such |
03:21.02 | dlynes_office | asterisk-dud: do you have asterisk up and running? |
03:21.03 | surfdue | like i have on my old server |
03:21.10 | surfdue | except i dont know how to cuase i use to use freepbx, which sucked. |
03:21.24 | surfdue | so im trying to move over 'the book' is very LONG :P |
03:21.32 | surfdue | ty for helping though |
03:22.30 | asterisk-dud | yes |
03:22.50 | asterisk-dud | dlynes_office: i get a dialtone from one channelbank and not from the other right now |
03:23.42 | [TK]D-Fender | asterisk-dud : pastebin your zaptel.conf |
03:24.08 | asterisk-dud | ok, i fixed |
03:24.14 | asterisk-dud | sorry to bother u guys |
03:24.33 | blitzrage | ! ! ! |
03:24.47 | dlynes_office | asterisk-dud: you had a misconfig on your channel bank? |
03:24.52 | blitzrage | !! !! !! |
03:25.03 | dlynes_office | !!! !!! !!! |
03:25.08 | surfdue | ! |
03:25.09 | surfdue | :P |
03:25.20 | [TK]D-Fender | blitzrage : I DON'T WANT RELATIONSHIP! |
03:25.33 | surfdue | [TK]D-Fender, i know you :P |
03:25.43 | blitzrage | lol |
03:25.59 | [TK]D-Fender | :| |
03:27.14 | blitzrage | |: |
03:27.35 | *** join/#asterisk focks (n=craig@74.130.97.237) |
03:29.23 | *** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane) |
03:29.24 | *** part/#asterisk trixter (n=trixter@65-165-167-217.du.volcano.net) |
03:32.13 | dlynes_office | surfdue: back up your existing file, and try this one: http://pastebin.com/761438 |
03:32.19 | dlynes_office | on that note, i'm out for a while |
03:32.26 | surfdue | k |
03:32.27 | surfdue | ty |
03:32.47 | *** part/#asterisk dlynes_office (n=dlynes@216.251.149.66) |
03:32.53 | *** join/#asterisk dlynes_office (n=dlynes@216.251.149.66) |
03:33.02 | dlynes_office | oops...wrong window :) |
03:35.26 | surfdue | hey same error |
03:35.26 | surfdue | :P |
03:38.06 | mitcheloc | dlynes_home: hi |
03:44.20 | Eric-xx | i have a question , i setup a incoming trunk and linked to a digital recp, everything works well when i call the incoming trunk |
03:44.38 | Eric-xx | but i can't seems to be able to see the incoming callerid/did number |
03:45.02 | Eric-xx | all i could see is -- Executing Set("SIP/651234567-7daa", "FROM_DID=s") |
03:45.14 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
03:45.26 | Eric-xx | anyone know's how i could see the callerid of a incomming route |
03:45.43 | [TK]D-Fender | Eric-xx : Please read the channel topic.... |
03:46.05 | Eric-xx | k |
03:47.03 | *** part/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
03:48.24 | *** join/#asterisk Winkie (n=urmom@cpc3-stre1-0-0-cust656.bagu.cable.ntl.com) |
03:51.31 | *** join/#asterisk bmg505 (n=leon@c1-161-9.rndf.isadsl.co.za) |
03:56.06 | hads | Does this look like something I should bring up here, or over in asterisk-dev land? http://pastebin.com/761623 |
04:00.08 | *** join/#asterisk PrOsHoCk (i=PrOsHoCk@ppp-71-140-3-39.dsl.scrm01.pacbell.net) |
04:11.29 | *** join/#asterisk znoG (n=gs@109-130-89-200.fibertel.com.ar) |
04:13.08 | surfdue | anyone? |
04:15.07 | litage | how do you determine which modules asterisk has loaded? |
04:15.38 | russellb | show modules |
04:16.59 | *** join/#asterisk terrapen_ (n=cjs@mauritius.island.nu) |
04:17.39 | litage | russellb: ``show modules'' lists all modules, not just the ones that are loaded |
04:17.41 | *** part/#asterisk terrapen_ (n=cjs@mauritius.island.nu) |
04:17.51 | *** join/#asterisk terrapen_ (n=cjs@mauritius.island.nu) |
04:17.52 | russellb | litage: it lists all the modules that are loaded. |
04:18.01 | litage | ah. thanks. didn't notice that part |
04:18.02 | litage | :) |
04:18.22 | russellb | asterisk doesn't know anything about modules not loaded :) |
04:22.48 | *** join/#asterisk sternn (n=sternn@user-0c938ku.cable.mindspring.com) |
04:23.39 | asterisk-dud | i want to make a dialplan that will take in the EXTEN argument, take that value and find the variable that is equal to EXTEN |
04:23.57 | asterisk-dud | so if EXTEN = 2030, and 2030 = ZAP/1 |
04:24.13 | asterisk-dud | then asterisk would dial zap/1 |
04:24.19 | asterisk-dud | if 2030 is dialed |
04:24.47 | asterisk-dud | i can't get it to take in the value of variable 2030 |
04:25.24 | asterisk-dud | exten => _20NX,1,Set(CHANNEL = ${EXTEN}) |
04:25.24 | *** join/#asterisk docelm0 (n=docelmo@55-65.126-70.tampabay.res.rr.com) |
04:25.46 | docelm0 | oi |
04:25.53 | asterisk-dud | now a want to take the value of channel and find the variable equal to it, and retrieve it's value |
04:26.07 | docelm0 | ~seen zoa |
04:26.10 | jbot | zoa <n=kkk@pirus.securax.be> was last seen on IRC in channel #asterisk, 11h 52m 12s ago, saying: 'you need a gsm plugin for winamp'. |
04:26.13 | asterisk-dud | and then dial that channel |
04:26.27 | docelm0 | crap just missed em |
04:29.57 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
04:30.55 | [shodan] | is this usable => http://www.i-rocks.com/2004/product/IR-2500.htm ? |
04:31.37 | *** part/#asterisk BenderNZ (n=bender@ip-58-28-96-39.wxnz.net) |
04:32.53 | *** join/#asterisk asterisk-dud (n=dwwollma@64-42-247-120.mb.skyweb.ca) |
04:33.07 | asterisk-dud | did anyone get the question i posted? |
04:33.15 | asterisk-dud | i get kicked off |
04:35.59 | asterisk-dud | any one out there? |
04:37.35 | kernel20 | hello |
04:38.08 | kernel20 | would it possible to put password to number that dials that starts on _951X |
04:38.09 | kernel20 | ? |
04:38.28 | kernel20 | if that number has been pressed it will asked for password |
04:38.40 | kernel20 | before a call can be stablished |
04:38.49 | asterisk-dud | lets say the value of variable CHANNEL is 2333, how can i take that and find variable 2333, and use it's valur |
04:40.31 | asterisk-dud | anybody? |
04:41.07 | *** join/#asterisk cryptnix (n=andrew@64.25.198.123) |
04:41.37 | asterisk-dud | anyone know a good SIP app for pocket pc |
04:41.42 | asterisk-dud | that will work with asterisk |
04:44.15 | *** join/#asterisk denon (n=denon@synapse.subneural.net) |
04:44.15 | *** mode/#asterisk [+o denon] by ChanServ |
04:51.38 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
05:01.46 | *** join/#asterisk Strom_C (n=strom@gateway.digium.com) |
05:02.28 | Strom_C | yo |
05:02.42 | asterisk-dud | yo |
05:02.57 | Strom_C | is there anyone in here who wouldn't mind sending me a test fax? |
05:03.20 | asterisk-dud | where r u? |
05:03.25 | Strom_C | Long Beach, California |
05:03.38 | asterisk-dud | long ways from here |
05:04.09 | Strom_C | asterisk-dud: canada counts as a long way away? :) |
05:04.18 | asterisk-dud | lol |
05:04.22 | Strom_C | you're in the same numbering plan space for crying out loud |
05:04.24 | asterisk-dud | what number, i can try |
05:04.41 | Strom_C | see privmsg |
05:05.02 | nick125 | if I had a fax machine plugged in, I wouldn't mind sending you one..but, i dont :( |
05:05.39 | surfdue | nick125, i do |
05:05.43 | surfdue | braodvoice.com |
05:05.52 | nick125 | surfdue: you didn't see the last requirement |
05:06.00 | surfdue | money order? |
05:06.02 | Strom_C | nick125: why the money order requirement? |
05:06.12 | nick125 | Strom_C: no credit card |
05:06.24 | Strom_C | get one of those prepaid credit cards tehn |
05:06.33 | nick125 | broadvoice prohibits them |
05:07.21 | Strom_C | well, if you absolutely insist on not using a credit card, then you could always go back to the 1950s |
05:07.31 | asterisk-dud | guys, if I have a variable 2030 = Zap/1, I have a dialplan that needs to take in EXTEN 2030 and return Zap/1 (the value of 2030) |
05:07.36 | asterisk-dud | how can i do that |
05:07.46 | surfdue | nick125, try lingo |
05:07.51 | surfdue | nick125, call them up |
05:08.08 | surfdue | whats a moneyorder again? |
05:10.56 | *** join/#asterisk freebsd_fan (n=ebola@i-83-67-73-117.freedom2surf.net) |
05:10.56 | nick125 | lol |
05:11.15 | surfdue | http://www.meritcall.com/neworderinfo.html |
05:13.13 | kernel20 | would it possible to put password to number that dials that starts on _951X |
05:13.20 | kernel20 | if that number has been pressed it will asked for password |
05:13.22 | kernel20 | before a call can be stablished |
05:13.23 | kernel20 | ? |
05:14.52 | surfdue | cya all |
05:16.19 | *** part/#asterisk sternn (n=sternn@user-0c938ku.cable.mindspring.com) |
05:18.07 | Strom_C | what do you know - faxing over voip works beautifully when you're only on a LAN :) |
05:21.17 | h3x0r | yewah |
05:21.27 | h3x0r | i had a customer doing 672 calls on a single pentium 4 of faxing |
05:23.33 | kaldemar | kernel20: take a look at application Authenticate. |
05:23.59 | *** join/#asterisk iceyp (n=icepick@firewall.unix.co.nz) |
05:24.33 | iceyp | hey guys, i just got 2 cisco 7940's with skinny client on them, im trying to update to P003-08-2-00 is it possible to do a dirct update or do i need a lower version first? |
05:24.40 | kernel20 | ok htanks |
05:24.46 | kernel20 | another one |
05:25.22 | kernel20 | i wanted to have all checking of voicemailmain start with #plus_sip_number |
05:28.11 | kernel20 | ie #11001 |
05:28.16 | kernel20 | it will check for voicemail |
05:28.18 | kernel20 | any ideas? |
05:29.09 | kernel20 | exten => #11001,2,Macro(mymail,11001@myvoicemail) |
05:29.12 | kernel20 | wont work |
05:29.18 | kernel20 | any ideas? |
05:31.54 | kaldemar | is that extension not working or your macro? |
05:32.25 | kernel20 | yeah |
05:33.42 | X-Rob | gwg |
05:33.43 | X-Rob | heh |
05:33.52 | kaldemar | yeah? |
05:33.58 | X-Rob | gotta love either/or questions that are answered by 'yeah' |
05:34.46 | kernel20 | yes |
05:34.55 | kaldemar | they're the best. |
05:36.10 | kernel20 | ? |
05:37.04 | kaldemar | - do you want to take the read pill, or the blue pill? - yeah. |
05:37.19 | kernel20 | both |
05:37.33 | kernel20 | anybody? |
05:37.36 | eipi | there's anyway to configure the iaxy to register into named server? or only by ip? |
05:37.57 | kernel20 | i want to have my voicemail checking that users should start #+their nummbert |
05:38.10 | kernel20 | ? |
05:38.12 | kaldemar | try sip debug to see if your phone is actually sending # to asterisk. a good way to start. |
05:38.19 | [shodan] | is this any good => http://www.i-rocks.com/2004/product/IR-2500.htm ? |
05:38.45 | kernel20 | i am on asterisk -vvvc now |
05:38.52 | kernel20 | i can see any logs pertaining it |
05:39.11 | kernel20 | i can't see any logs pertaining it |
05:39.12 | dlynes_home | CunningPike: almost got one of those blood sangoma cards working now :p |
05:39.17 | dlynes_home | s/blood/bloody/ |
05:39.27 | CunningPike | dlynes_home: Having fun yet? |
05:40.11 | dlynes_home | CunningPike: are you kidding? ie still can't do png alpha layers? |
05:40.16 | kernel20 | i want to have my voicemail checking that users should start #+their nummber |
05:40.19 | kernel20 | any ideas? |
05:40.24 | kernel20 | dlynes_home:? |
05:40.26 | CunningPike | dlynes_home: Nope - not with out a custom filter |
05:40.38 | dlynes_home | that's pretty ghey |
05:40.45 | CunningPike | dlynes_home: Ya think??? |
05:40.55 | dlynes_home | png's have been a standard for how long now? |
05:41.00 | kernel20 | is this correct |
05:41.01 | kernel20 | exten => _#11001,2,Macro(nkymmail,11001@nkymvoicemail,#11001) |
05:41.07 | CunningPike | dlynes_home: Since the flood - Noah had some |
05:41.08 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
05:41.13 | kernel20 | exten => ${ARG2},1,VoiceMailMain(${ARG1}) |
05:41.24 | dlynes_home | man |
05:41.27 | dlynes_home | i need to buy more dishes |
05:41.29 | kernel20 | ? |
05:41.33 | harryvv | dishes? |
05:41.39 | CunningPike | dlynes_home: TMI ;) |
05:41.40 | dlynes_home | I keep running out of clean dishes all the time :p |
05:41.42 | kernel20 | ? |
05:42.02 | harryvv | ohh I thought it was your wife that was PMsing on you and thoughing the dishes at you. |
05:42.08 | kernel20 | > |
05:42.09 | kernel20 | ? |
05:42.13 | dlynes_home | nah...not married, thankfully |
05:42.16 | harryvv | :) |
05:42.29 | dlynes_home | otherwise i'd have been pushed off a cliff a long time ago :p |
05:42.41 | harryvv | ahh yea woman are easy to anger |
05:42.44 | kernel20 | ? |
05:42.52 | kernel20 | ? |
05:42.53 | kernel20 | ? |
05:42.58 | dlynes_home | yeah...i feel sorry for cunningpike |
05:43.04 | CunningPike | kernel20: There's something wrong with your keyboard |
05:43.06 | harryvv | Ive been bussy all day and now im resting with a ache body and a headache. |
05:43.12 | dlynes_home | he's irish, so i would imagine his wife is too :) |
05:43.15 | CunningPike | dlynes_home: You've never met my wife ;) |
05:43.28 | CunningPike | dlynes_home: 10 years married last week |
05:43.29 | harryvv | dlynes your in van right? |
05:43.34 | dlynes_home | CunningPike: ah...your wife's not irish? |
05:43.40 | CunningPike | dlynes_home: Yes, she is |
05:43.51 | dlynes_home | So she doesn't have a nasty temper? |
05:44.14 | dlynes_home | I always hear all this stuff about how irish woman have nasty tempers :) |
05:44.19 | dlynes_home | s/woman/women/ |
05:44.34 | kernel20 | ? |
05:44.35 | dlynes_home | harryvv: correct |
05:44.35 | kernel20 | ? |
05:44.35 | harryvv | I knew a irish male that had a temper |
05:44.36 | kernel20 | ? |
05:44.37 | LoRez | Warning: `kernel20' seems to be spamming, please discontinue or kills/klines will be issued. |
05:44.37 | Rez | Warning: `kernel20' seems to be spamming, please discontinue or kills/klines will be issued. |
05:44.37 | kernel20 | ? |
05:44.43 | kernel20 | ? |
05:44.51 | dlynes_home | lol |
05:44.53 | dlynes_home | it's about time :) |
05:44.58 | harryvv | her wife danced with a rock hudson type and he steamed. Then the divorse followed. |
05:45.03 | dlynes_home | I had him ignored, so I didn't even notice he was spamming :) |
05:45.05 | harryvv | his wife I mean |
05:45.19 | mitcheloc | *sigh* AND D-Fender told me to be nice to him earlier.... |
05:45.26 | dlynes_home | mitcheloc: lol |
05:46.00 | dlynes_home | he's apparently banned from most of efnet |
05:46.18 | dlynes_home | CunningPike: i didn't even think you were that old yet |
05:46.27 | dlynes_home | CunningPike: ten years married...damn |
05:46.31 | mitcheloc | lol, if he made it into that many chat rooms you'd have thought he'd pick up something? |
05:46.35 | harryvv | I am a little worried about this terrorist capture. Some times I work at key fedeal sites on the side that are likly targets here in BC. |
05:47.08 | CunningPike | dlynes_home: Yup - 10 years. I'll be 40 next year |
05:47.17 | *** join/#asterisk rvhi (n=rv@66.175.65.89) |
05:47.20 | dlynes_home | CunningPike: damn...didn't think you were anywhere near that old |
05:47.29 | dlynes_home | CunningPike: thought you were early to mid 30's |
05:47.29 | mitcheloc | dlynes_office: happy snappin? |
05:47.38 | harryvv | anyway night all |
05:47.41 | CunningPike | harryvv: Much more likely to be a) run over in a crosswalk b) shot by a gangsta |
05:47.52 | CunningPike | dlynes_home: Well, thanks :D |
05:47.53 | *** join/#asterisk denon (n=denon@synapse.subneural.net) |
05:47.53 | *** mode/#asterisk [+o denon] by ChanServ |
05:48.04 | dlynes_home | lol |
05:48.16 | dlynes_home | mitcheloc: not until you make it compatible with linux :) |
05:48.57 | dlynes_home | mitcheloc: and even then, i'd probably only get limited use out of it |
05:49.06 | dlynes_home | mitcheloc: my boss would probably get a lot of use out of it though |
05:49.59 | mitcheloc | dlynes_home: well hell, get your boss to use it, and i'll get it to work on linux, good deal? |
05:50.30 | dlynes_home | heh |
05:50.49 | dlynes_home | mitcheloc: btw, do you have post dial dtmf support? |
05:51.20 | mitcheloc | it was just added to asterisk managment api |
05:51.30 | mitcheloc | i'll add it in, i need to figure out a clean interface though |
05:51.54 | mitcheloc | there are so many features i have to add, it's going to get stuffy looking =( |
05:52.26 | dlynes_home | so you need to have access to the asterisk manager socket, for that to work? |
05:52.36 | mitcheloc | yes sir |
05:52.53 | dlynes_home | so if the server is only listening on the localhost, you won't be able to interface |
05:53.05 | mitcheloc | no sir, you'll have to fix that |
05:53.34 | dlynes_home | well, what i'm saying is that it's a no go on my main server then |
05:53.40 | dlynes_home | but all the office pbxes it would be fine for |
05:54.16 | mitcheloc | okay, well, a step at a time, i'm working on improving it |
05:54.21 | dlynes_home | are you able to lock the administrator settings on the dialer? |
05:54.26 | *** join/#asterisk postel (n=jp@unaffiliated/postel) |
05:54.31 | dlynes_home | i.e. through a password protect? |
05:55.36 | mitcheloc | well manager api already requires a username/password |
05:55.39 | mitcheloc | is that what you mean? |
05:55.43 | mitcheloc | but it needs to be on the client |
05:55.46 | mitcheloc | like an instant messenger |
05:56.09 | dlynes_home | mitcheloc: no...i meant the settings for the client to know which server to connect to, what port, what server for the manager api, what port for the manager api, all that kinda stuff |
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05:56.43 | dlynes_home | mitcheloc: a lot of people running asterisk don't want their customers to know its asterisk |
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05:58.15 | mitcheloc | dlynes_office: are you referring to where it says "Asterisk Management API"? |
05:58.26 | mitcheloc | i could come up with something to disguise that... |
05:59.02 | dlynes_home | mitcheloc: yeah, but also we don't usually like to have the end user monkeying with the settings, either |
05:59.20 | dlynes_home | once they start doing that then we have to come down and play tech support guys |
05:59.31 | dlynes_home | not fun |
05:59.49 | mitcheloc | i understand, i've gotten mixed reactions on that, some people like it, others don't, i'm thinking of a hybrid way to load and lock |
06:00.06 | mitcheloc | sort of, user types in username/password, settings downloaded and locked in, and the user can't see or modify them |
06:03.37 | dlynes_home | mitcheloc: or maybe an oem toolkit to allow you to customize it? |
06:03.50 | dlynes_home | mitcheloc: including slapping your own logos on it? |
06:04.28 | mitcheloc | dlynes_home: i work with resellers, all you have to do is let me know what you need and we can work something out |
06:04.39 | mitcheloc | also, there is a deploy example i worked on, so you can set up a logon script |
06:04.50 | dlynes_home | ah |
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06:07.42 | mitcheloc | dlynes_office: can you read this, and try the download at the bottom? http://www.snapanumber.com/Support/Forums/tabid/58/forumid/1/threadid/69/scope/posts/Default.aspx |
06:07.45 | mitcheloc | let me know what you think |
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06:09.24 | dlynes_home | mitcheloc: is that a windows extension or is it the linux version? |
06:09.35 | mitcheloc | which extension? |
06:09.44 | dlynes_home | deploysnap.zip |
06:09.56 | mitcheloc | .zip is a zip file? |
06:10.08 | mitcheloc | heh, but i don't think thats your question? |
06:10.31 | dlynes_home | deploysnap.zip is for use on windows or linux? |
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06:10.58 | mitcheloc | it's for use on windows |
06:11.01 | dlynes_home | ah |
06:11.04 | dlynes_home | the reason i'm asking |
06:11.07 | mitcheloc | like on a logon script, it's only an example, you could do it 100+ ways |
06:11.14 | dlynes_home | is because i don't have access to a windows box here |
06:11.35 | dlynes_home | all my machines at home are either linux or solaris |
06:11.38 | mitcheloc | well, just open the zip and look inside the logon.bat file, it should explain what you are asking |
06:11.52 | dlynes_home | ah...that's what you were pointing me to then |
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06:15.58 | dlynes_home | mitcheloc: yeah...pretty simple stufff |
06:16.17 | dlynes_home | mitcheloc: it could be a lot more elegant in perl though, and then you wouldn't need sqlite.exe :) |
06:16.18 | mitcheloc | it's just an sql database file |
06:16.29 | mitcheloc | yep, well, so you have part of your answer |
06:16.42 | mitcheloc | i was thinking of a name/value called locked/yes and hidden/yes |
06:16.53 | mitcheloc | so that it can't be updated in snap or seen (depending on yes/no) |
06:17.08 | mitcheloc | of course any scripter can do this, but most corporate users can't...it's a quick fix for now |
06:17.29 | dlynes_home | mitcheloc: well, most of our customers are small offices |
06:17.48 | dlynes_home | the chance of one person in the office having a clue enough to go digging into the registry is pretty remote |
06:18.02 | dlynes_home | and even if they did, they probably know enough to know what they're doing with the settings |
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06:19.22 | mitcheloc | dlynes_home: very true, so is there anything missing then to help make this deployable *now*? |
06:19.36 | mitcheloc | (btw, there is a very cool auto-update feature in snap, so you don't have to maintain it on your client's computers) |
06:20.45 | dlynes_home | cool...have you used amsn? |
06:21.09 | dlynes_home | none of the stuff i mentioned would stop it from being deployable |
06:21.35 | dlynes_home | they were just things that are pretty much close to being a necessity for offices |
06:21.35 | mitcheloc | amsn? or msn? i use msn heh |
06:21.36 | dlynes_home | amsn |
06:21.48 | dlynes_home | It's a TK msn messenger client |
06:21.57 | dlynes_home | it runs on linux |
06:22.04 | dlynes_home | it has an autoupdate feature, too |
06:22.06 | mitcheloc | ah, wikipedia entry, yep, no i don't, why? |
06:22.13 | mitcheloc | ah, cool |
06:22.14 | dlynes_home | ~wiki amsn |
06:22.30 | dlynes_home | yep...same amsn |
06:22.32 | mitcheloc | i use trillian, i just hover over "amsn" and it tells me ;), i thought it was a typo though |
06:23.26 | dlynes_home | and unlike msn where you have to wait a while before hitting nudge |
06:23.35 | dlynes_home | with amsn you can hit it repeatedly as fast as you like |
06:23.41 | dlynes_home | heh |
06:23.45 | mitcheloc | lol, nice feature |
06:24.15 | dlynes_home | good for those clowns that start talking to you, want you to help them, and then seem to forget they were talking to you |
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06:26.07 | mitcheloc | ;), so you pointed out amsn to tell me it updates? =P |
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06:31.25 | dlynes_home | yeah :0 |
06:31.54 | dlynes_home | Kiraly Parcina! |
06:33.48 | mitcheloc | kiraly parcina? |
06:34.14 | dlynes_home | hungarian |
06:34.47 | dlynes_home | kiraly is hungarian for king :p |
06:35.08 | tparcina | dlynes, i'm not from hungary, i'm from croatia! :)) |
06:35.16 | dlynes_home | oh yeah :((( |
06:35.18 | mitcheloc | take that! |
06:35.32 | dlynes_home | His tld keeps throwing me off :(( |
06:35.37 | tparcina | but in ast one year i have been for three weeks in budapest, so maybe it counts :)) |
06:36.20 | tparcina | and i find hungarian girls weary sweet and nice, who nows maybe i'll become hungarian :)) |
06:36.45 | dlynes_home | hahaha |
06:37.04 | dlynes_home | i dunno |
06:37.13 | dlynes_home | i've seen the odd croatian chick that's pretty hot, too |
06:37.25 | tparcina | how are you? is it around midnight at your place? |
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06:37.41 | tparcina | what old croatian chick? |
06:37.42 | dlynes_home | 11:30pm, yeah |
06:37.49 | dlynes_home | odd, not old |
06:38.02 | mitcheloc | dlynes_office: don't lie, you like old chicks |
06:38.26 | dlynes_home | what do you consider old, mitch? |
06:38.45 | mitcheloc | anything over 30 is over the hill |
06:39.01 | dlynes_home | ah then yeah...i definitely like old chicks |
06:39.02 | tparcina | yes, i like one old cihck alsoo - 22 years old :) |
06:39.21 | dlynes_home | i like chicks around mid 30's or so |
06:39.29 | drray | women are like dog crap, the older they get, the less messy it is to pick them up |
06:39.36 | mitcheloc | heck, 16-30, thats my limit |
06:39.38 | mitcheloc | ** 18 |
06:39.44 | tparcina | as you go older the border young/olod is moving... |
06:40.03 | mitcheloc | drray: nice |
06:40.38 | tparcina | yes drray, i have never heard that one before, it's good |
06:41.01 | dlynes_home | yeah..younger chicks have a lot more problems |
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06:42.08 | tparcina | yes, they have problems with themself! |
06:42.48 | dlynes_home | yeah...where as old guys just play with themself! |
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06:44.27 | mitcheloc | tparcina: like this... mv -rf / /dev/null |
06:44.27 | dlynes_home | lol |
06:44.35 | mitcheloc | dlynes_office: ygpm ;) |
06:44.42 | dlynes_home | ygpm? |
06:44.48 | mitcheloc | private message |
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06:54.13 | tparcina | Mitcheloc, when you know so much about disk formating :)) give me a hand with this one |
06:54.25 | tparcina | when i installed FC4 i have formated two hdd with disk druid. both disk head two partitions, first one from 1-3916 and another from 3917-4569. then i taked out one disk, remouved all partitions with partition magic and then i have put it back to fedora machine. then i have formated disk with fdisk and i tried to create the same partitions but i have failed. this is what fdisk showes me - http://pastebin.ca/62450 so, how to format disk so it's the sa |
06:54.56 | mitcheloc | tparcina: it was a joke, i'm heading to sleep though, sorry |
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06:56.33 | P-NuT | Hi all, has anyone got an x100p working under ubuntu? |
06:57.04 | tparcina | mitcheloc, i know it was a joke, that why i have put :)) - anyway, you can sleep tomorow :)) |
06:57.31 | drray | it's already tomorrow here |
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07:20.29 | littleball | hello, i don't think it is a good idea to store all sip user configuration in sip.conf file. if there are 10,000 users, the file will be too big. |
07:20.36 | littleball | and not easy to maintainence |
07:20.44 | littleball | any suggestion? |
07:21.43 | syle | mysql realtime |
07:26.54 | littleball | syle, thanks. another questions about the deployment of media relay system. because most of sip phones will be behind firewall, i am planing to use a few asterisk box as media relay server. refer to http://mit.edu/sip/sip.edu/started.shtml |
07:27.13 | littleball | Architecture figure |
07:27.59 | parag7732 | Can anybody suggest small billing System for a small company ???? |
07:28.04 | littleball | assuming all sip phones are behind firewall. and a few asterisk media gateway is needed. |
07:30.24 | parag7732 | Can anybody suggest small billing System for a small company ???? |
07:30.29 | littleball | what is exact mean of "nat=yes" in sip.conf ? |
07:35.54 | qdk_ | parag7732: have you looked at voip_info.org? |
07:36.15 | parag7732 | yes qdk |
07:36.17 | parag7732 | I tried |
07:36.21 | parag7732 | a2billing |
07:36.23 | parag7732 | is there |
07:36.30 | parag7732 | but that is prepaid and postpaid billing |
07:36.34 | *** join/#asterisk Shaun2222 (n=ndci@ip68-5-63-223.oc.oc.cox.net) |
07:36.36 | parag7732 | I need simple billing system |
07:36.47 | parag7732 | for a small company for mobile phone usage |
07:37.21 | qdk_ | littleball: it means that your SIP-phone is able to traverse the NAT its behind. |
07:37.22 | Shaun2222 | are there any linksys routers or anything with built in stun servers? |
07:38.08 | qdk_ | Shaun2222: are the any routers with buildin STUN servers? |
07:38.22 | Shaun2222 | i dont know |
07:38.31 | qdk_ | parag7732: what is the besides pre and post billing? |
07:38.40 | qdk_ | Shaun2222: i would say no. |
07:39.13 | parag7732 | like if in the company people calls to mobile or landline I need to add Rate field |
07:39.35 | Shaun2222 | i have one of those zyxel p-2000w phones, it's going to be behind a NAT where the SIP server is on the public network. |
07:40.23 | qdk_ | parag7732: make a simple script parsing the CDR.csv fil and the dump whatever you wish to bill. |
07:40.28 | Shaun2222 | it says for nat it supports outbound proxy, stun or manual configured wan/sip add |
07:41.50 | *** join/#asterisk dec (n=tom@ppp169-75.lns3.adl4.internode.on.net) |
07:42.27 | qdk_ | Shaun2222: A STUN server i located somewhere on the internet, probably on the same network as your the * you register to. |
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07:42.49 | Shaun2222 | whats the point of that. |
07:43.12 | Shaun2222 | i need these phones to work inside a home users little crap NAT network |
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07:43.26 | Shaun2222 | connecting to a * server out on a public ip |
07:43.33 | qdk_ | Shaun2222: then dont use STUN? |
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07:43.46 | Shaun2222 | qdk_: i though that was needed for NAT? |
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07:43.59 | ghenry | Hardware and Asterisk version for 5 9's uptime? |
07:44.05 | ghenry | Anyone at that level? |
07:44.12 | qdk_ | Shaun2222: no, i have a lot of SIPs behind nat without any problems. |
07:44.36 | Shaun2222 | qdk_: so do i but they are all cisco 7960 phones |
07:44.38 | ghenry | brb |
07:44.40 | qdk_ | ghenry: oone can tell you... dependS!!!! |
07:44.48 | Dico_ | is anybody knows the difference between 'hint' and ' device status' ? |
07:44.51 | Shaun2222 | this little zytel thingy doesnt appear to work |
07:45.10 | Shaun2222 | i also remember somebody says the cisco phones had some built in stun server or somthing |
07:45.11 | qdk_ | Shaun2222: ok, i dont have any Cisco equ. |
07:45.33 | Shaun2222 | i had to tell the cisco phones that they where on a NAT |
07:45.49 | Shaun2222 | for them to work |
07:45.50 | Dico_ | oups. sorry, i forget to say 'hello' ^^. My apologize : hello world :) |
07:45.58 | qdk_ | Shaun2222: i dont know much about STUN, but im pretty sure that it doesnt make sense to have the STUN behind the NAT. |
07:46.06 | Shaun2222 | ok |
07:46.42 | ghenry | Ha, I thought so qdk_ |
07:46.54 | qdk_ | Dico_: hehe... in what situation are you talking about? |
07:46.58 | ghenry | Was looking at IBM BladeCenter T |
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07:47.32 | Dico_ | humm for the 'away' , 'busy' and so on, using subscribe and notify |
07:47.35 | ghenry | is it worth getting the enterprise version from digium or just using the OSS one? I would have thought they would be near enought the same |
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07:48.37 | qdk_ | ghenry: well the specific hardware is only a matter of load and how many you have i your cluster to make sure of the 5 9's. |
07:48.46 | Dico_ | qdk_, actually, at first i thought hint and device status were the same stuff, but after having a look in manager.c, it appears to be a difference ... |
07:49.01 | ghenry | qdk_: yup |
07:49.22 | qdk_ | Dico_: ok, dont know the difference. sorry. |
07:49.31 | Dico_ | ok |
07:50.03 | qdk_ | ghenry: you could also use different cards... |
07:50.09 | Dico_ | where is my oej ? ^^ |
07:50.30 | RoyK | -- --- .-. -. .. -. --. |
07:50.37 | RoyK | Dico_: prolly still sleeping |
07:50.47 | qdk_ | ghenry: i would say that a strong setup is made up of different hardware and a large "cluster" solution. |
07:51.18 | Dico_ | RoyK, ok |
07:51.20 | RoyK | five nines and asterisk is probably like riding a $100 bike in 200Mph |
07:52.02 | qdk_ | RoyK: in a SPoF setup, yes. |
07:52.03 | ghenry | qdk_: Thanks |
07:52.20 | ghenry | SPoF? |
07:52.32 | qdk_ | Single Point of Failure |
07:52.55 | RoyK | sure, but wtf will you put in the front to gain five nines? |
07:53.06 | qdk_ | robl^: infront? |
07:53.15 | ghenry | doh, thanks |
07:53.25 | qdk_ | RoyK: ups, for you. ;-) |
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07:54.40 | qdk_ | RoyK: I would probably make a BSD with CARP setup, so i would have 1 IP for a many *'s, providing failover and/or loadbalancing. |
07:56.05 | qdk_ | my current setup is 2 single and independet frontend servers with automatic failover to and from my peer and to and from my backend servers. |
07:56.18 | RoyK | qdk_: I have an idea, no (or almost no) code yet, but take a look: http://sitsotd.org/ |
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07:58.25 | zoa | royk, i actually have that |
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07:58.28 | zoa | running in production |
07:59.35 | zoa | well kinda anyway |
08:00.07 | BugKham | how to silently reload asterisk? |
08:00.13 | RoyK | zoa: you do? |
08:00.16 | zoa | yes |
08:00.26 | zoa | i have such a thing for a year |
08:00.33 | zoa | its part of my asterisk cluster |
08:00.40 | zoa | need to go to meeting now |
08:00.46 | zoa | and no i will not give it to you :) |
08:00.57 | zoa | its part of our isp asterisk cluster solution |
08:01.04 | dec | I haven't been following development too closely, but how is the PLC going for SIP? Anything stable available yet? |
08:01.07 | zoa | that will probably never see the day of light |
08:01.14 | zoa | dec: its in trunk |
08:01.32 | dec | zoa - stable enough for production? |
08:02.52 | zoa | yes |
08:02.56 | zoa | well depends |
08:02.58 | zoa | try it and see |
08:03.04 | dec | OK. :) |
08:04.27 | qdk_ | So sitsotd removes the problem of lost sessions, when a server(in a cluster) goes byebye? |
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08:06.38 | RoyK | zoa: btw, it might seem the new patches from slav works |
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08:21.16 | parag7732 | Failed to execute SQL : SQL CREATE TABLE rate_engine_rate ( rate_id integer primary key auto_increment, route_id integer NOT NULL references rate_engine_egress(route_id), type varchar(3), country varchar(40) NOT NULL, extra varchar(40), prefix varchar(10) NOT NULL, active_date date, expires_date date, firstperiod integer NOT NULL, periods integer NOT NULL, startcost float, periodcost float, trialcost float, ); failed : You have an error in your SQL synta |
08:21.43 | parag7732 | help please |
08:22.01 | macTijn | does this look like #sql ? |
08:22.15 | parag7732 | why |
08:22.31 | parag7732 | got it |
08:23.48 | Greek-Boy | since my D-Link DES-1526 is not providing power to my cisco phones (7912 and 7960), can I make my own custom patch panel and provide power to it? |
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08:29.14 | bmg505 | I do that with wrt54GL routers, 2 wires, just be carefull of the polarity |
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08:34.30 | Greek-Boy | which wires though? |
08:35.18 | RoyK | doesn't matter |
08:36.46 | opus_ | hello |
08:36.50 | opus_ | hello |
08:36.58 | opus_ | is there a built in unix backtrace() command? |
08:39.02 | stephane_ | re |
08:40.22 | RoyK | opus_: gdb asterisk core.xxxx and then 'bt' |
08:40.36 | opus_ | RoyK I want to do it programmically |
08:41.39 | RoyK | why? |
08:42.34 | opus_ | debugging a deadlock in of course a multithreaded app |
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08:50.04 | Greek-Boy | so u guys are saying i must put power on any 2 wires to power up the cisco phone? |
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08:56.07 | RoyK | opus_: i thought gdb could do that as well |
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08:56.46 | RoyK | opus_: thead apply all bt |
08:56.48 | RoyK | iirc |
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08:59.03 | Greek-Boy | where can i get the PoE pinouts for cisco 7912? |
09:00.36 | x86 | stephane_: re |
09:02.03 | tparcina | I have done it without those guys in #fedora or ##linux channel :)) software RAID-1 works for me :))) |
09:02.48 | tparcina | Greek-Boy: as far as i know, cisco 7912 uses prestandard PoE, and if you want to use it you'll have to buy cisco PoE switch |
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09:03.25 | tparcina | have you discuss anything interesting while i was building RAID? |
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09:04.33 | Greek-Boy | tparcina does that countf or the 7960 too? |
09:06.51 | tparcina | Greek-Boy: I think so. but 7961 uses standard PoE, so it can be used with any poe switch |
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09:09.24 | tparcina | Greek-Boy: to tell you the thrue, i have 1 7905 that works with standard POE and 6 7905/7912 that doesn't. and i have one 7920 that works with standard POE and 4 7940/7960 that doesn't. so, there is (at least) one hardware revision that works with POE, but most of them don't. |
09:09.56 | tparcina | so, if you would like to use 7905, 7912, 7940 and 7960 on POE, buy cisco POE switch. |
09:10.09 | Greek-Boy | Yeah but I hard dumb terminals power them up |
09:10.17 | tparcina | and if you would like to use cisco phones on any other POE switch, then buy 7941 or 7961 |
09:10.21 | Greek-Boy | so i thought about building my own supply as a quick solution |
09:11.03 | *** join/#asterisk sgnome` (i=sgnome@ACC0F21A.ipt.aol.com) |
09:11.24 | tparcina | i have counted, and it would cost me more to buy non-cisco POE switch + terminals (or to buy cisco poe switch) then to buy power cubes for phones. so i bouth power cubes |
09:11.51 | tparcina | and power cubes work for sure :)) |
09:12.17 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
09:12.49 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
09:13.13 | *** join/#asterisk speedwagon (n=Ariel@dsl-20-177.cofs.net) |
09:16.29 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
09:21.48 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
09:25.02 | SheriF_WorK | can anyone point me to a doc about SIP protocol and how it works ? |
09:25.33 | littlejohn | SheriF_WorK, read the rfc |
09:25.42 | maagic | 4th match from google: http://en.wikipedia.org/wiki/Session_Initiation_Protocol |
09:25.47 | Makenshi | http://www.faqs.org/rfcs/rfc3261.html |
09:26.05 | sgnome` | SIP Demystified is an excellent book on the topic from Addision Wesley |
09:26.08 | key2 | SheriF_WorK: SIP RFC |
09:26.10 | SheriF_WorK | cool thx guys |
09:27.40 | SheriF_WorK | i just need to know exactly how SIP works .. for ex i have an asterisk and 2 users registering to asterisk from 2 countries and they are talking right now ... so that all the traffic is passed throw asterisk ? or asterisk just opend the port and give infromation to both clients and they open some ports between eachothers and working without needing asterisk help anymore after the session is initiated ? |
09:30.39 | RoyK | http://comics.linuxzealot.net/Dilbert/2006.06.06.gif |
09:39.37 | *** join/#asterisk zagaya972 (n=d2s-comp@APointe-a-Pitre-102-1-3-9.w81-248.abo.wanadoo.fr) |
09:42.55 | tparcina | RoyK, stop joking around, this is serius channel :)) |
09:43.19 | tparcina | btw, RoyK, do it, do it, do it :)) |
09:46.10 | key2 | When asteirsk is in realtime mode |
09:46.11 | clive- | i hear they just released asterisk business versi0n supporting aculab and dialogic crads on asterisk |
09:46.20 | key2 | if I do a reaload, I lose all the queue member, is it normal ? |
09:46.22 | Dico_ | oej, hello; sorry for annying you. I had a look in manager.c . Can you tell me the difference between 'hint' and 'device status' ? Is it the same thing a different level ? |
10:00.15 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
10:00.31 | *** join/#asterisk AltnTab (n=ecs@nrjsoft13.networx-bg.com) |
10:03.35 | *** join/#asterisk frk2 (n=faraz@202.5.145.13) |
10:10.26 | frk2 | dudes- my grandstream has the screen blanking problem..... is this supposed to be normal? |
10:18.46 | kmilitzer | Is there a way to disallow SIP-REFER-requests in asterisk? |
10:26.32 | Dico_ | ok, nvm |
10:26.34 | Dico_ | cu all |
10:30.25 | *** join/#asterisk mmmmmToop (n=mmmmToop@firewall.datapro.co.za) |
10:33.38 | clive- | mike |
10:35.54 | *** join/#asterisk lorinc (n=ang@caracas-1720.adsl.interware.hu) |
10:38.21 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
10:43.14 | *** join/#asterisk viperdude (n=jon@84.45.193.6) |
10:44.08 | mmmmmToop | Hi Clive.. |
10:45.32 | fourcheeze | how do I get one asterisk to dial another and take itself out of the loop? I've been looking at Transfer() but that doesn't seem to work for me |
10:46.00 | viperdude | fourcheeze: allow reinvites |
10:46.11 | opus_ | not sure, but in theory you can use chan_sip's reinvite feature with transfer to get out of the loop |
10:46.23 | fourcheeze | viperdude: does that mean I have set the second * as a user? |
10:46.43 | fourcheeze | ATM I'm just trying Transfer(SIP/ext@host) |
10:46.49 | viperdude | erm not sure I think it will also work with peer |
10:46.53 | fourcheeze | that just seems to die |
10:47.14 | opus_ | thats asterisk! |
10:47.19 | opus_ | why do you think they call it asterisk |
10:47.40 | fourcheeze | because it's what you explete when it drops calls? |
10:47.47 | opus_ | exactly |
10:48.01 | *** join/#asterisk eenduik (n=albert@sthwuha.fh-reutlingen.de) |
10:48.08 | fourcheeze | well, I'm about 95% happy with it's reliability so far |
10:48.22 | opus_ | yeah |
10:48.35 | fourcheeze | so if I was to put in a normal dial() and allow reinvites on the destination |
10:48.42 | fourcheeze | then it should lose the first * ? |
10:49.17 | viperdude | should do if the calling party excepts reinvites |
10:49.39 | fourcheeze | ok, I'll give it a go |
10:50.00 | fourcheeze | now assuming this works - how many calls per second can a single * handle like this? |
10:50.12 | *** join/#asterisk cybergypsy (n=mark@APoitiers-156-1-10-74.w86-207.abo.wanadoo.fr) |
10:50.14 | opus_ | 4-5 |
10:50.18 | fourcheeze | or rather how many should I leave it before starting to think of using SER |
10:50.34 | fourcheeze | 4-5 a second will probably do |
10:50.35 | opus_ | depending on your application |
10:51.15 | fourcheeze | suppose I needed a * to stay in the middle (i.e. the calling party didn't accept reinvites) how many of those could * pass through at once? |
10:52.13 | opus_ | just passing a call through maybe 100-200 |
10:52.30 | opus_ | doing something with that call like IVR , i wouldn't trust it with more then 80 calls |
10:52.31 | fourcheeze | so that's not really an option for scaling |
10:52.31 | *** join/#asterisk jgoo (n=e4b80e21@foodtecsolutions.com) |
10:52.55 | opus_ | even on the fastest systems. it will use only like 2% cpu, but most of the code is to fucked up and not threadsafe |
10:52.58 | fourcheeze | opus_: how did you guess that I'm building an IVR box? |
10:53.05 | *** join/#asterisk s-ndh-c (i=michi@gw2.routing.tuxhost.de) |
10:53.17 | opus_ | you should hire methen:) |
10:53.21 | opus_ | i build large scale ivr systems |
10:53.24 | fourcheeze | aha |
10:53.28 | zoa | royk, all he did was disable debugging logging |
10:53.28 | fourcheeze | how much do you charge? |
10:53.35 | zoa | so that might have made the difference for you |
10:53.59 | opus_ | fourcheeze depending on the type of work of couse |
10:54.07 | opus_ | fourcheeze and currency ? |
10:54.45 | fourcheeze | Yen? |
10:54.50 | fourcheeze | Rupee? |
10:54.53 | fourcheeze | Rouble? |
10:54.58 | jgoo | Coffee beans? |
10:55.05 | opus_ | i try to be just under the best people but there is always some dumb ass that comes up with the $3k bid that I have to compete with. |
10:55.16 | opus_ | USD |
10:55.32 | fourcheeze | to be honest this isn't going to be large-scale particularly |
10:55.58 | jgoo | I got an email from colognechip regarding BRI cards, if anyone wants a copy |
10:56.10 | jgoo | links, info, drivers, very nice email |
10:56.11 | zoa | jgoo |
10:56.12 | zoa | gimme |
10:56.26 | jgoo | msg me email, ill forward |
10:56.29 | fourcheeze | I just want to divert incoming calls to an IVR box and then back into the main system |
10:56.39 | opus_ | fourcheeze, simple enough |
10:56.42 | *** join/#asterisk eset (n=eset@ip545186e3.direct-adsl.nl) |
10:57.25 | eset | hi, are there any really basic web interfaces for creating/appying for sip accounts for *? |
10:58.13 | opus_ | i gotta go to bed, but i will PM you my email if you want to jump start your project |
10:58.24 | fourcheeze | opus_: should be fine as long as I can actually get a transfer to work! |
10:58.51 | opus_ | canreinvite is also a global setting , even though it isn't documented anywhere :) |
10:59.01 | opus_ | canreinvite=yes at the top of sip.conf will be respected |
10:59.06 | opus_ | in my case I wanted canreinvite=no |
10:59.09 | opus_ | take it easy |
10:59.13 | fourcheeze | ok, l8rz |
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11:19.03 | MrChimpy | bugger. when i trigger a setvar from my AGI script it doesn't produce any AMI event - presumably because the one you'd monitor is Newexten and because I'm using AGI to do the SetVar I'm not actually in a dialplan |
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11:54.42 | ness | brettnem: I tried the "sip show user" today - doesn't work (no such command). I managed to do it (more or less) by using voicmemousers, but it is a bit hacky... |
12:00.27 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
12:00.47 | key2 | someone has tryed the queue realtime ? for some reason, after 30sec, the user in queue gets a hangup |
12:00.49 | key2 | any idea ? |
12:01.08 | *** join/#asterisk aze (n=aze@ACayenne-101-1-12-98.w81-248.abo.wanadoo.fr) |
12:04.43 | s-ndh-c | what hardware would i need to route calls between asterisk and my existing pbx? |
12:05.07 | s-ndh-c | i guess if i use a single isdn line i can only transfer 2 calls at the same time right? |
12:07.16 | *** join/#asterisk fenlander (n=fenlande@82.152.81.57) |
12:08.41 | *** join/#asterisk twisla (i=twisla@lutin.jard.in) |
12:09.38 | *** join/#asterisk myiagy (n=myiagy@mail.voffice.com.br) |
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12:14.03 | key2 | s-ndh-c right |
12:17.18 | s-ndh-c | so i would need some special card or something , cause i need atleast 4 calls at the same time between existing pbx and asterisk |
12:17.27 | s-ndh-c | better more |
12:17.30 | s-ndh-c | :) |
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12:20.20 | *** join/#asterisk coppice (n=chatzill@44.199.17.210.dyn.pacific.net.hk) |
12:21.47 | eipi | there's anyway to make iaxy connect thru a name server to asteriks? |
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12:25.15 | *** part/#asterisk tparcina (n=tparcina@wr-lama.iskon.hr) |
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12:29.09 | RoyK | hm. how is an e1 loopback plug wired? |
12:32.05 | *** part/#asterisk ness (n=Tom@pppin-4e-b6.pop-kaltenengers.rz-online.NET) |
12:32.28 | I-MOD | RoyK, pin 1 to pin 4 and pin 2 to pin 5 |
12:32.59 | I-MOD | eipi, iaxys dont do DNS resolution, IP addresses only |
12:33.04 | key2 | RoyK: the RJ45 ? |
12:34.06 | *** join/#asterisk Kis (i=vlad@p5080D187.dip.t-dialin.net) |
12:35.24 | RoyK | yeah |
12:35.26 | RoyK | found out |
12:35.40 | RoyK | I-MOD: thanks |
12:35.45 | I-MOD | np |
12:37.27 | kdz13|gone | so then, SIP fixes all my dtmf issues |
12:41.04 | *** join/#asterisk Ariel_ (n=Ariel@70.46.87.158) |
12:41.28 | key2 | RoyK: do you know why when I have a user that gets into the queue, if he stays more than 30sec, he gets a hangup |
12:41.28 | key2 | ? |
12:42.25 | RoyK | because of timeout=30 ? |
12:43.34 | key2 | RoyK: I have a timeout = 300 |
12:43.37 | key2 | but still |
12:43.45 | key2 | RoyK: it's in realtime |
12:46.03 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
12:46.32 | RoyK | key2: dunno. set debug = lots |
12:47.01 | MrChimpy | is there some existing grouping mechanism or should I be implementing this at the dialplan or AGI level? |
12:47.02 | [TK]D-Fender | MrChimpy : Shove the dial lines back to back. |
12:47.16 | MrChimpy | what do you mean tkd? |
12:47.24 | MrChimpy | oh, in the dialplan? |
12:47.43 | MatsK | RoyK: Its not RJ45 its RJ48, http://www.arcelect.com/RJ48C_and_RJ48S_8_position_jack_.htm |
12:47.45 | [TK]D-Fender | MrChimpy : just put the steps back-to-back. If it connects on the first it'll never hit the 2nd after a hangup, and all is good. On failure of first, who cares about the reason? Just dial the 2nd. |
12:47.50 | [TK]D-Fender | MrChimpy : Yes. |
12:48.28 | *** join/#asterisk aze (n=aze@ACayenne-101-1-12-98.w81-248.abo.wanadoo.fr) |
12:48.29 | MrChimpy | ok, cool. that'll do for dialplan apps. I'll do something fancier in AGI. |
12:49.42 | [TK]D-Fender | MrChimpy : Its 1 line! Would you like your elephant gun sir? |
12:49.43 | coppice | RJ48C is just RJ45, with the 5 crossed out, and 8C written in crayon |
12:49.44 | *** join/#asterisk Bert- (n=bert@LAubervilliers-151-12-81-84.w193-252.abo.wanadoo.fr) |
12:49.47 | Bert- | hello there |
12:50.06 | key2 | RoyK: where do you set debug=lost? |
12:50.17 | MrChimpy | tkd: this is a 1000 line IVR system. elephant gun required. |
12:50.31 | I-MOD | cept RJ48C can accept 4, 6, and 8 wire connectors while a RJ45 can only accept 8 wire connectors |
12:50.35 | RoyK | key2: at the console, set verbose 9, set debug 9, and try again |
12:50.45 | MrChimpy | needs to be as resilient as I can make it |
12:50.51 | RoyK | key2: make sure you log debug info somewhere (logger.conf) |
12:52.45 | Bert- | lol |
12:53.05 | Bert- | will Next versions of asterisk support the new cpu instruction 'maybe' ? |
12:53.06 | Bert- | huhu :) |
12:53.17 | [TK]D-Fender | MrChimpy : Big deal... I had 347 worth of STDEXTEN macro and associated setup contextx for it :) |
12:54.02 | Bert- | It will very funny to see malloc() returning 'maybe' :d |
12:54.19 | SheriF_WorK | [TK]D-Fender: hey man ;-) ... i have something in mind and i need u to show me the light .. now about the SIP protcol it's self .. the media packets and the all the talking packets is send directly between the to endpoint bypassing the SIP server " asterisk " right ? |
12:55.00 | I-MOD | SheriF_WorK, only if you allow reinvites |
12:55.12 | [TK]D-Fender | SheriF_WorK : Depends.... SIP always passes through the server, RTP is variables (canreinvite) |
12:55.17 | *** join/#asterisk Killa200 (n=killa200@adsl-153-147-238.cha.bellsouth.net) |
12:56.02 | SheriF_WorK | [TK]D-Fender: if my softphones in another NAT networks " each end in a NAT " i should use canreinvite = no .. other way it will not work. |
12:56.20 | SheriF_WorK | i'm reading in the RFC for like 2 hours and my brain is damged already :D |
12:56.36 | [TK]D-Fender | SheriF_WorK : No reinvites "or else" ..... |
12:57.08 | SheriF_WorK | ? |
12:57.38 | [TK]D-Fender | SheriF_WorK : describ the full path of the call like : phone -> nat -> internet -> * -> nat -> otherphone |
12:58.10 | SheriF_WorK | so canreinvite = yes " the media and the RTP will bypass the asterisk " canreinvite = no the RTP will pass throw the asterisk ? |
12:58.14 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
12:58.45 | [TK]D-Fender | SheriF_WorK : Correct. Only if set on BOTH ends of the call and assuming conditions allow |
12:58.55 | SheriF_WorK | oh ok phone -> nat -> internet -> * <-internet <- nat <- otherphone |
13:00.00 | [TK]D-Fender | SheriF_WorK : Forget reinvites... thats not going to work. |
13:00.30 | *** join/#asterisk Katty (n=angela@64.82.232.54) |
13:00.58 | SheriF_WorK | [TK]D-Fender: i didn't get it sorry :-s what will not going to work ? this setup works with me using canreinvite = no .. |
13:01.32 | [TK]D-Fender | SheriF_WorK : Thats what I was saying... it will not work WITH reinvites |
13:01.54 | *** join/#asterisk Shoragan (n=shoragan@134.169.175.72) |
13:02.01 | SheriF_WorK | ahh yes :-) |
13:02.02 | SheriF_WorK | hehe |
13:02.03 | SheriF_WorK | sorry |
13:02.26 | SheriF_WorK | [TK]D-Fender: so with this way all the trafic and the call trafic will pass throw * ? |
13:03.34 | *** join/#asterisk zotz (n=zotz@24.244.133.115) |
13:03.43 | frk2 | Grandstream issues again!!! this time my screen blanks and i get no voice... dialout to an extension works though :( |
13:04.03 | frk2 | Even my Cisco hangs!!!! Cisco 7960G using SIp 8.2 |
13:04.07 | kdz13 | is the voip-info wiki the best source of AGI documentation |
13:04.12 | [TK]D-Fender | SheriF_WorK : Yup, everything will flow through *. |
13:04.14 | kdz13 | or is there another more complete source? |
13:04.35 | [TK]D-Fender | kdz13 : Wiki + The Book |
13:04.38 | [TK]D-Fender | ~book |
13:04.39 | jbot | i guess book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
13:05.38 | kdz13 | ah sweet |
13:05.43 | kdz13 | i was not aware of this book. |
13:05.47 | *** join/#asterisk feld_ (n=feld@12.148.212.157) |
13:06.06 | kdz13 | [TK]D-Fender: thanks |
13:06.13 | [TK]D-Fender | kdz13 : np, good luck |
13:06.49 | *** join/#asterisk epablo (n=epablo@WLL-24-pppoe194.t-net.net.ve) |
13:07.04 | feld_ | is there a common reason for registration error timeouts with X-Lite? |
13:07.12 | SheriF_WorK | [TK]D-Fender: thx ;-) |
13:07.30 | epablo | Hi people |
13:07.43 | feld_ | hi epablo |
13:07.58 | [TK]D-Fender | feld : Usually bad user/pass |
13:08.17 | *** join/#asterisk zeppelin_ (n=zeppelin@201-40-157-135.paemt700.dsl.brasiltelecom.net.br) |
13:08.25 | Katty | morning |
13:08.32 | epablo | Anyone with some experience with queues? |
13:08.54 | *** join/#asterisk Arno[Slack] (n=root@66-163-12-60.ip.tor.radiant.net) |
13:09.07 | coppice | i'm british. I've had a lifetime of studying queues |
13:09.45 | kdz13 | coppice: lol |
13:09.51 | zeppelin_ | :D |
13:09.52 | sevard | I've found conflicting articles. I'm betting that voip-supply is out of date... Does asterisk have T.38 support ? |
13:09.53 | epablo | I need to run a command in order tu signal my CRM to open a popup on the agents screen before I pass him a call |
13:10.45 | epablo | sevard: it supports it, but I have read that it doesn't garantie it |
13:10.50 | feld_ | aha! nevermind people. somehow my nat=true disappeared from sip.conf |
13:10.52 | feld_ | silly me |
13:10.53 | feld_ | :) |
13:11.03 | sevard | epablo: is that why the bounty is still 10,750? |
13:11.21 | coppice | sevard: some t.38 support is going into 1.4. Right now * doesn't support T.38 |
13:12.01 | epablo | I have used * to send faxes and most of the times it has worked.. |
13:12.10 | epablo | maybye it was pure luck ;) |
13:12.15 | [TK]D-Fender | coppice : Isn't that just T.38 PASS-THROUGH support? |
13:12.36 | feld_ | anyone have any tips on how to make it so that when an extension is dialed that doesnt exist the caller is notified of it? |
13:12.39 | coppice | yeah, but I have the rest pretty much complete :-) |
13:12.53 | coppice | it won't go into SVN, though |
13:13.07 | [TK]D-Fender | coppice : Community owes you a lot.... and I think that 10G would go a fair ways ;) |
13:13.15 | [TK]D-Fender | coppice : Disclaimer issues? |
13:13.28 | coppice | yes. the T.38 code in in spandsp |
13:13.46 | sevard | I have an ATA hooked up to * and I'm sending calls over a PRI, it looks like faxs are going through and come back in but I'm getting "Unknown RTP codec 100 received" which apparently means I'm using T.30 and faxes are not supposed to be working at all |
13:13.57 | coppice | yes, i think 10 giga dollars will do nicely |
13:14.22 | [TK]D-Fender | coppice : Sorry.. typo... "g" ;) |
13:14.39 | kdz13 | ~book |
13:14.40 | jbot | book is probably a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
13:15.05 | mut | uh |
13:15.14 | mut | anyone know where to get a cheap prism 2 pcmcia card? |
13:15.16 | coppice | Unknown RTP codec can mean many things. you need to look at the SDP to see what 100 was allocated as |
13:15.22 | mut | searching on ebay is needle in a hat stack style |
13:15.59 | sevard | coppice: I'm not sure what SDP is. |
13:16.31 | coppice | session dementing protocol. it the stuff in the SIP where the codec config was set up |
13:16.45 | *** join/#asterisk gandhijee (n=gandhije@mail.win-ent.com) |
13:17.04 | sevard | So you want me to do a sip debug and look for.. what? |
13:17.19 | sevard | or zap debug |
13:17.31 | [TK]D-Fender | sevard : SIP debug on the neg |
13:17.31 | feld_ | im having nat issues apparently. i can register and dial but i cant seem to receive any calls.... :( any tips? |
13:17.43 | [TK]D-Fender | feld : pastebin your SIP peer entry |
13:17.44 | [TK]D-Fender | ~pb |
13:17.46 | jbot | somebody said pb was a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
13:18.10 | feld_ | [TK]D-Fender, only x-lite phones at the moment |
13:18.10 | *** join/#asterisk dpryo (n=hn@raphael.ondskap.net) |
13:18.11 | sevard | [TK]D-Fender: say what brutha |
13:18.29 | sevard | feld_: he's asking for your sip.conf |
13:18.48 | feld_ | sevard, heh i know :) im pasting... :P |
13:18.52 | sevard | coppice: so... want to explain that a bit more for me?:) |
13:19.31 | feld_ | http://sh.nu/p/1923 |
13:19.48 | coppice | do SIP debug. look at the call setup, and you should see numbers assigned to various codecs. look for what is assigned as 100 |
13:20.19 | [TK]D-Fender | feld : Add "canreinvite=no", and "qualify=yes" to your peer entries. Is your * behind NAT as well by any chance? |
13:20.37 | sevard | Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP |
13:20.59 | feld_ | [TK]D-Fender, no my * is on a real IP |
13:21.54 | [TK]D-Fender | feld : Then add what I just suggested, do a "sip reload", and restart your X-Lite's and all should be good. |
13:22.00 | sevard | coppice: is that what I'm looking for?> |
13:22.22 | eipi | how i can configure my iaxy to register a named server? |
13:23.20 | epablo | I need to signal my CRM to open a popup on the agents screen before I pass him a call. Is this posible with * queues? |
13:24.33 | *** join/#asterisk feld_ (n=feld@12.148.212.157) |
13:24.43 | feld_ | crap didnt mean to close that... hehe |
13:25.41 | viperdude | hi gus |
13:25.45 | feld_ | [TK]D-Fender, hrm still no good. i'm dialing from the * console for testing and it can't seem to reach my phone yet... let me paste my little extensions.conf. maybe something is wrong with it. I'm fairly new to *.... |
13:25.55 | gandhijee | anyone know if there is a crosstools channel on this server? |
13:25.56 | viperdude | is there a way to set the privacy flag on outgoing SIP calls? |
13:26.22 | feld_ | http://sh.nu/p/1924 |
13:26.39 | techman97_andy | hey all, when was the original Asterisk project started? (year?) |
13:26.48 | feld_ | techman97_andy, i think 98? |
13:26.51 | techman97_andy | cool |
13:26.53 | techman97_andy | thx |
13:27.04 | viperdude | the copyright on the CLI starts at 1999 |
13:27.07 | feld_ | at least i think thats what the oreilley pdf said |
13:27.27 | *** join/#asterisk philippel (n=p_lindhe@c-24-19-186-72.hsd1.wa.comcast.net) |
13:28.54 | *** join/#asterisk miztic (n=gerard@rarcoa.com) |
13:29.07 | [TK]D-Fender | feld_ : Your dialplan is a nice idea, and would work except that your x-lites have NAMES for their [] entry, and not numbers. |
13:29.14 | astar` | hello i want to do shortcuts on numbers : ex : i type 25 on my phone and its dials 0243434343 |
13:29.45 | [TK]D-Fender | feld : You'd need to be able to do like "Dial(SIP/markxlite)", which "x" does NOT catch..... |
13:29.49 | feld_ | [TK]D-Fender, i borrowed chunks of the dialplan from somewhere else. Thanks for solving it though :) |
13:29.54 | viperdude | aster`: exten =>25,1,Dial(SIP/0243434343@myvoipproider,20,tr) |
13:30.23 | feld_ | I should have known that. I guess I just didnt piece it together and realize the variable was failing. :) |
13:30.35 | astar` | actually i have something like that but its doesn't respect the dial because its dials directly ONe trunk |
13:30.46 | [TK]D-Fender | feld_ : np, mod it up and let me know. |
13:30.55 | [TK]D-Fender | feld : 2 second finx if you put your mind to it ;) |
13:30.56 | astar` | dial *rules |
13:31.21 | viperdude | aster`: not sure what you mean |
13:31.32 | sevard | feld_: how are you dialing ? |
13:31.48 | s-ndh-c | what am i doing wrong? |
13:31.50 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.239.142.Dial1.SanJose1.Level3.net) |
13:31.54 | [TK]D-Fender | sevard : I've ID'd the problem... its a minor fix I'm sure he's already working on. |
13:32.04 | sevard | I'm just curious how he's dialing |
13:32.17 | s-ndh-c | i i call myself using xlite i get a missed call where the address is asterisk@myproxyip |
13:32.20 | feld_ | [TK]D-Fender, hehe yeah thanks. I appreciate your help. I'm definitely sticking around in this channel as I'll be using this for a long time.... at work, selling, implementing, and supporting it. So I'll be around to help others, too :P |
13:32.26 | *** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.239.142.Dial1.SanJose1.Level3.net) |
13:32.57 | [TK]D-Fender | feld : If that escaped you and is a sample of the kind of work you're doing, you MIGHT want to study a bit more ;) |
13:33.05 | s-ndh-c | why does it send asterisk@myproxyip as the adress? |
13:33.18 | s-ndh-c | not my username or my extension@myproxyip? |
13:33.27 | [TK]D-Fender | s-ndh-c : My guess is you didn't set CallerID for the calling channel. |
13:33.31 | *** join/#asterisk kaz0358 (n=kaz@kazg5.telecom.ksu.edu) |
13:33.38 | s-ndh-c | [TK]D-Fender: where do i do that? |
13:34.03 | feld_ | sevard, i was dialing from the console. currently my context is local so it works if i just do "dial #" instead of doing "dial #@context" |
13:34.11 | epablo | I once used a command called chan_grab, but I don't remember where I downloaded it. Does anyone know I can find it? |
13:34.28 | sevard | feld_: I don't have the "dial" command |
13:34.36 | feld_ | sevard, strange |
13:34.48 | sevard | [TK]D-Fender? |
13:35.24 | MatsK | the dial command is only present if the sound card support is loaded in asterisk |
13:35.34 | sevard | I see. |
13:35.40 | [TK]D-Fender | s-ndh-c : in your sip.conf peer entries for all related phones. |
13:35.53 | feld_ | oh yeah it uses OSS or ALSA (ugh...). |
13:36.02 | sevard | whowhat I love alsa :) |
13:36.53 | feld_ | sevard, i know some game programmers... epic, ID, etc... and they've told me that the ALSA devs themselves have admitted that their API is horribly broken and that for some devices the frequencies generated by ALSA are way off. |
13:37.10 | *** join/#asterisk mercestes (n=merceste@69.15.174.114) |
13:37.23 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
13:37.49 | feld_ | ALSA is open source = good. Broken and stubbord devs = bad. I like 4Front's OSS, but it's not open source. It works well, though. Pretty powerful actually. :) |
13:38.03 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
13:38.03 | *** mode/#asterisk [+o anthm] by ChanServ |
13:38.20 | feld_ | "to each his own" that's the linux way and that's what matters ;) |
13:38.21 | s-ndh-c | ok |
13:39.04 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
13:40.00 | feld_ | [TK]D-Fender, any suggestions how to throw in logic so that if an unknown extension is dialed it will play a diff audio msg instead of my simple straight-forward dialing plan I have now? |
13:40.44 | epablo | Last year the was like an asterisk branch. Does anyone remember the name of that project? |
13:41.14 | MikeJ[Laptop] | yep |
13:41.23 | feld_ | epablo, freeswitch |
13:41.34 | feld_ | it's pretty alpha though they're claiming otherwise. =S |
13:41.52 | MikeJ[Laptop] | freeswitch isn't an asterisk branch |
13:41.56 | epablo | <PROTECTED> |
13:42.07 | feld_ | MikeJ[Laptop], it's not? i swear it is? |
13:42.11 | MikeJ[Laptop] | nope |
13:42.15 | MikeJ[Laptop] | totally new code |
13:42.23 | epablo | It was a rebelion.. LOL |
13:42.32 | MikeJ[Laptop] | that was openpbx |
13:42.40 | MikeJ[Laptop] | it;s dead as far as I know. |
13:42.49 | *** join/#asterisk mosty (i=mostynm@60-241-198-194.static.tpgi.com.au) |
13:42.53 | feld_ | oh yes you're correct MikeJ[Laptop] |
13:42.56 | [TK]D-Fender | feld : don't use a generic mask like that. Hard code th valid ones. |
13:42.58 | feld_ | my bad :P |
13:43.17 | feld_ | [TK]D-Fender, ok and then what do I throw in there to pick up the rest of the extensions? |
13:43.36 | MikeJ[Laptop] | epablo, what app you looking for/. |
13:43.38 | [TK]D-Fender | feld : you don't... they'll just 404 like they should |
13:43.47 | feld_ | [TK]D-Fender, beautiful. thanks. |
13:43.52 | epablo | MikeJ[Laptop]: You are right.. Thanks |
13:44.03 | MikeJ[Laptop] | epablo, what app are you looking for? |
13:44.45 | mosty | i need to edit /etc/modprobe.d/zaptel to make sure opermode is set to AUSTRALIA, but there is a line at the top of this file that says it's automatically generated and not to edit it. so what generates this file? |
13:45.50 | MikeJ[Laptop] | sounds like you are using one of the gui's? |
13:47.55 | feld_ | My boss and another cow-orker who is not usually available threw this * box together. It does have a Digium TDM400P in it with 4ports but I dont know which are FXO or FXS. Any way via software to find that out? |
13:47.56 | *** join/#asterisk znoG (n=gs@109-130-89-200.fibertel.com.ar) |
13:47.58 | epablo | MikeJ[Laptop]: chan_grab |
13:48.37 | wunderkin | epablo, it is app_changrab |
13:49.00 | MikeJ[Laptop] | epablo, www.pbxfreeware.org is it's home. |
13:49.48 | epablo | MikeJ[Laptop]: Thanks |
13:49.50 | MikeJ[Laptop] | it probably needs some updating for 1.4 if your not using it on stable... |
13:49.51 | astar` | someone know if there is something to dial a phone number via egroupware ? |
13:50.04 | MikeJ[Laptop] | if so, let me know, I can get the updates posted up there |
13:50.49 | feld_ | omg |
13:50.55 | MikeJ[Laptop] | ? |
13:50.56 | feld_ | music on hold works by default? |
13:51.04 | feld_ | lol nice somehow it is..... |
13:51.05 | *** join/#asterisk dools (n=iain@125.62.65.184) |
13:51.20 | feld_ | i called myself, picked up the second line. it essentially put myself on hold and I hear the music. |
13:51.30 | epablo | Ok.. thanks |
13:51.32 | *** join/#asterisk Creperum (n=ilya@tex.tsua.net) |
13:51.35 | MikeJ[Laptop] | did they finally change default moh in the sample configs to use native?? |
13:51.42 | dools | hi, is there anything in agi like "register termination handler" or something like that so that i can execute some code after everything else has happend? |
13:52.24 | Creperum | hey, how can i make FOP to display 100 extentions and 60 trunks???? |
13:52.24 | mosty | feld: you could open the case and look, of just try all the combinations for zaptel.conf (there's not that many) |
13:52.40 | Creperum | it only displays part of them! |
13:53.33 | feld_ | mosty, i'd rather open it up. I'll do that. not like this thing can go into production soon anyway. idiots moved this thing live without DNS, i have not been given our T1's DNS info after I've been asking for the last 3 days, and * isnt updated to fix the vulnerability =S |
13:54.27 | mosty | feld: well the two types of modules will most likely be different colors, then you have only two possible options for zaptel.conf |
13:54.39 | stephane_ | re |
13:55.21 | mosty | run ztcfg and it will spit out errors if you get it wrong |
13:57.00 | dools | okay, perhaps some more background information ... i'm hacking a reseller module into a2billing and i've set up a 'resource allocation' model which means that when a user makes a call, it calculates the total number of minutes possible and 'allocates' those resources from the reseller's credit so that a reseller is not able to have his user's cost us money. i can quite happily allocate the resources at the beginning of the exeuction of a2biling.php, but the s |
13:57.35 | dools | otherwise i don't know where to de-allocate the resources |
13:58.09 | wunderkin | feld_, you can use any dns server |
13:58.27 | *** join/#asterisk C4T3l (n=rcall01@216.54.143.2) |
13:59.24 | feld_ | wunderkin, not usually. most will block by who they are servicing in the case of ISPs. |
13:59.35 | techman97_andy | anyone worked with a specific Cisco SIP phone? Any recommendations on a standard desk set? |
13:59.42 | feld_ | occasionally you can find some open ones but I never have any luck doing that :( got any suggestions? |
14:00.23 | dools | techman97_andy: i use cisco ATA, works well, config interface is ugly as hell (ie. i wouldn't want to support it unless i shipped it pre-configured) |
14:00.38 | C4T3l | techman97_andy: we use the Cisco 7960 series... thy're ok phones |
14:01.00 | dools | techman97_andy: cisco ATA allows you to plug in a standard phone + a LAN cable |
14:02.23 | Vorondil | hi all; quick question: shouldn't this (http://pastebin.com/762326) play music on hold songs in random order? it always seems to play them alphabetically. =/ |
14:03.02 | techman97_andy | cool all - thanks! |
14:03.04 | dools | Vorondil: maybe you're just really really unlucky |
14:03.06 | mosty | vorondil: prefix the filenames with a random X digit number? |
14:03.28 | *** join/#asterisk Ecio (n=eciostar@194.105.59.42) |
14:03.34 | Ecio | hi |
14:03.34 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
14:04.36 | Ecio | does anybody know if it's possible to use the new IM and Presence features of X-lite 3.0 with asterisk ? |
14:05.23 | vader-- | hola |
14:06.11 | Vorondil | lol, mosty: i thought about that, but the point is that no caller gets the same song twice during, say, a week |
14:06.24 | *** join/#asterisk A-Tuin|work (n=A-Tuin@212.41.185.81) |
14:06.29 | Vorondil | since most folks never stay on hold for an entire song anyway |
14:06.45 | C4T3l | Ecio, I imagine if you use hints in the dial plan it sould work...?? |
14:07.07 | dools | Vorondil: seriously... who cares? |
14:07.19 | dools | Vorondil: it's hold music |
14:07.40 | Ecio | C4: yes im tryin, i've just read that publish is not supported so no extended features (like away etc..) |
14:08.00 | Ecio | btw Instant Messages doesnt seem to work... |
14:08.21 | Ecio | i see in debug: WARNING[16727]: chan_sip.c:7281 receive_message to X from Y, dropped it... |
14:08.30 | mosty | vorondil: just put loads of music in there and don't restart asterisk very often? |
14:08.41 | dools | haha shittest solutions ever |
14:09.02 | Vorondil | dools: eh, i mean, it's not a big deal, i just expected random=yes to do what i wanted. it didn't. so i asked. |
14:09.23 | dools | Vorondil: just submerge your server in jelly and play checkers for 9 hours |
14:09.31 | dools | Vorondil: i know, i'm being facetious |
14:09.38 | Vorondil | mosty: it seems to stop playing stuff (and rightfully so) when nobody's on hold, so it just starts over from the top again |
14:09.52 | Vorondil | lol |
14:10.02 | C4T3l | Ecio: asterisk will support text sent to a phone from the dialplan, but to my knowledge it will not support UA2UA Instant msging |
14:10.05 | Vorondil | the jelly and checkers does sound appealing.. |
14:10.09 | dools | Vorondil: maybe random=yes means that it randomly plays the hold music for a particular session |
14:10.31 | dools | Vorondil: is there are 'persistent hold music' option or something that would keep the same track going for all callers? |
14:10.34 | Ecio | c4 i see :( |
14:10.36 | *** join/#asterisk Hmmhesays (i=negative@66.173.103.110) |
14:11.05 | dools | Vorondil: what i know about asterisk, by the way, you could fit on the head of a pin plus 9 angels, so don't listen to me |
14:11.10 | *** join/#asterisk viler (i=1000@200.114.70.228) |
14:11.10 | C4T3l | Ecio: maybe someday it will :D |
14:11.15 | dools | Vorondil: i came here to ask a question about a2billing :-) |
14:12.01 | Vorondil | dools: hehe, s'all good. i hadn't touched asterisk until about a month ago |
14:12.42 | Vorondil | dools: but yeah, i thought the same thing about random=yes, so i listened all the way through a song, and it just went to the next alphabetic one |
14:12.52 | Vorondil | (assuming "alphabetic" is the right word there...) |
14:13.35 | *** join/#asterisk anonymouz666 (i=anonymou@200.218.196.5) |
14:13.44 | *** join/#asterisk lorinc (n=ang@caracas-2783.adsl.interware.hu) |
14:13.51 | dools | Vorondil: i must say, though, that i've spent hours and hours on hold, and the only thing i ever hear is the same cd repeated ad nauseum (in some cases literally) or a local radio station |
14:14.05 | *** join/#asterisk VoicePulse (n=contact@unaffiliated/voicepulse) |
14:14.22 | *** join/#asterisk b00mer (i=fwuser@blackhole.c5i.com) |
14:15.04 | Vorondil | dools: indeed, i'd like to keep from that if i can help it |
14:15.20 | dools | Vorondil: you're a nobler person than i :-) |
14:15.24 | vader-- | hehe |
14:16.00 | Vorondil | wait, so is "=" interchangeable with "=>" in the conf files? (namely musiconhold) |
14:16.17 | C4T3l | Ecio: i have an idea... Maybe you could try a canreinvite =yes in sip.conf, maybe you could set up 2 softphones and try it that way? *shrugs* |
14:16.32 | Ecio | c4t: that's what im doin :) |
14:16.48 | *** join/#asterisk aze_ (n=aze@ACayenne-101-1-8-198.w81-248.abo.wanadoo.fr) |
14:16.49 | frk2 | dudes- i have a question for a fellow grandstream GXP2000 user/sufferrer - anybody? |
14:16.51 | Vorondil | voip-info has musiconhold with =>'s |
14:16.51 | Vorondil | http://tinyurl.com/bn57o |
14:17.25 | frk2 | guys- why do hard phones Hang??? In my problematic client's network- even the Cisco 7960G hangs |
14:17.38 | drray | power? |
14:17.40 | frk2 | GXP 2000 works WAY better if I do POE on it... so points towards power issues. |
14:17.58 | drray | my ciscos don't hang |
14:18.04 | drray | which sip image? |
14:18.06 | frk2 | drray- thats what i think too |
14:18.07 | frk2 | 8.2 |
14:18.10 | frk2 | what u using? |
14:18.17 | drray | Im stuck in 6 |
14:18.19 | drray | er, 7 |
14:18.25 | *** join/#asterisk websae (n=websae@h69-129-251-26.69-129.unk.tds.net) |
14:18.31 | drray | I did not renew my smartnet contract |
14:18.34 | drray | :) |
14:18.36 | frk2 | maybe 7 is the way to go :) |
14:19.05 | frk2 | The crap with Cisco is that 802.3af is unknown to them, till recently |
14:19.07 | drray | do you have a power cube? |
14:19.27 | drray | I'd try a phone with a power cube, then blame the injector |
14:19.59 | frk2 | yup.. i have a power cube- thats what causes the issues |
14:20.13 | drray | oh |
14:20.17 | frk2 | See my grandstream GXP would hang DAILY/Hourly with local power |
14:20.53 | frk2 | so I got a 802.3af injector and hooked it up.. now only the screen blanks after 5 days |
14:20.53 | drray | hmmm |
14:20.53 | drray | dirty power? |
14:20.53 | frk2 | so its a huge improvement |
14:20.57 | frk2 | thats why im guessing |
14:21.07 | drray | run a phone on a UPS? |
14:21.13 | frk2 | However, ATcOM or Dlink shit phones work completely okay |
14:21.45 | frk2 | however |
14:21.58 | drray | maybe a drop to 7.3 (or whatever the highest 7.x is) is the way to go for one or two |
14:22.01 | frk2 | more interesting is that this 'hanging' paradigm only happens in a particular area in the office |
14:22.06 | Hmmhesays | blahbitty blahblah |
14:22.13 | Hmmhesays | buildroot for uclibc is pissing me off this morning |
14:22.26 | drray | that's external |
14:22.28 | frk2 | The cisco and the grandstream are in cubicles next the each other |
14:22.46 | drray | 50 foot extension cord |
14:22.46 | drray | :) |
14:24.35 | *** part/#asterisk dools (n=iain@125.62.65.184) |
14:25.38 | [TK]D-Fender | Polycom > All |
14:25.48 | jarrod | hey |
14:26.00 | jarrod | is there a way to make it so sip reload doesnt send a signal to all my polycoms |
14:26.04 | mut | well |
14:26.13 | drray | I don't (or did not at this time last year) like hoe Polycom's look |
14:26.18 | mut | polycom >= all |
14:27.01 | mut | anyon recommend a good prism 2 pcmcia card? |
14:27.44 | tzafrir | donno. I'm using a zd1211-based usb stick |
14:28.56 | mut | i have one of those, gf is using it in her pc tho |
14:29.17 | Hmmhesays | i was reminded last night why I was a bachelor for so long (speaking of girlfriends) |
14:29.28 | mosty | i heard that intel cards are currently the best supported right now |
14:29.47 | mut | mosty: looking for something i can use for sniffing |
14:30.17 | *** join/#asterisk swytch (n=ezcall@LNeuilly-152-22-86-193.w193-251.abo.wanadoo.fr) |
14:30.35 | feld_ | hey guys my voicemail is failing login because it keeps claiming the context=default. where in the voicemail extension line do I define the context from which the users calling it will be belonging to? |
14:31.24 | swytch | question about the manager api. specific events seem to miss from the output, like Event: Dial. is there a way to enable such events? |
14:31.48 | *** part/#asterisk epablo (n=epablo@WLL-24-pppoe194.t-net.net.ve) |
14:31.49 | jarrod | felds_: VoicemailMain(@context) ? |
14:32.26 | jarrod | swytch: i believe you can enable what output you want to see, or a verbosity of the default |
14:33.38 | mosty | how can i prevent ztdummy from being loaded? |
14:33.43 | feld_ | jarrod, i actually did (_X.@context) |
14:33.45 | swytch | jarrod: i tried "read = Dial" in manager.conf, but nothing. do i have to let asterisk dump a lot just to get the few events i want? |
14:33.49 | feld_ | just figured it out :) |
14:34.06 | *** join/#asterisk eKo1 (n=bernd@190.4.7.90) |
14:34.36 | *** join/#asterisk Joshaidan (n=icechat5@thunderbay-voip-4.vianet.ca) |
14:34.57 | eKo1 | I'm getting this NOTICE message on the Asterisk CLI since I connected my second PRI line this morning: chan_zap.c:8207 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 2 |
14:35.10 | *** join/#asterisk }btorch{ (n=kvirc@adelphi.geofocus.com) |
14:35.22 | drray | do you have a d channel on span 2? |
14:35.29 | eKo1 | Yep. |
14:35.46 | eKo1 | Or atleast, there should be. |
14:36.01 | drray | is it coming out of a channel bank? |
14:36.37 | drray | I had a similar issue, where everything about the PRI worked |
14:36.43 | drray | just kept getting a bad d channel |
14:37.11 | eKo1 | The PRI is comming out of a signaling gateway |
14:37.24 | eKo1 | that does SS7<->PRI conversion. |
14:38.25 | eKo1 | I've been googling around and, from the info. I've found, it could be an issue with the card. |
14:39.58 | jarrod | have another PRI card tot est with? |
14:41.43 | coppice | eKo1: you probably have a clock sync problem in your config. |
14:42.05 | eKo1 | OK. I think I know why it is all messed up. I botched my /etc/zaptel.conf |
14:42.54 | eKo1 | If the span is going to by the clock source, I should have span=1,0,0,ccs,hdb3,crc4 right? |
14:43.09 | coppice | nope |
14:43.25 | eKo1 | should it be span,1,1 then? |
14:43.38 | mosty | i have a wctdm card that isn't by digium, do i still need ztdummy? |
14:43.41 | coppice | if you want to take clock from the span it needs to be something like span=1,1,0,ccs,hdb3,crc4 |
14:44.19 | eKo1 | coppice: and if the clock is going to be obtained from the other end, it should be span=1,2 right? |
14:44.48 | coppice | nope. the 1 means the first priority source |
14:45.35 | eKo1 | What does that mean exactly? |
14:45.49 | *** join/#asterisk fholmes (n=fholmes@rrcs-24-227-237-197.sw.biz.rr.com) |
14:46.13 | vader-- | can anyone explain to me what this dialplan will do |
14:46.14 | vader-- | exten => _NXXXXXX,1,Dial(ZAP/g1/${EXTEN:1}) |
14:46.14 | vader-- | exten => _NXXXXXX,2,Congestion |
14:46.14 | vader-- | exten => _NXXXXXX,102,Congestion |
14:46.15 | tzanger | eKo1: it means that if that span is up, the card will sync to that span's clock. |
14:46.21 | *** join/#asterisk Ahrimanes (n=michael@62.61.133.90.generic-hostname.arrownet.dk) |
14:46.25 | fholmes | Does anyone here use Queues? I am just curious to find out if they have any problems with them I should know about before I spend time trying to implement them... |
14:46.26 | coppice | if you have a line coming in from an SS7 source you certainly want to slave to its timing. use span=1,1,0,ccs,hdb3,crc4 |
14:46.50 | tzanger | think of it as priorities. #1 = my preferred source. #2 = my next preferred source if #1 is gone, etc. #0 = fuck you, I will never sync to your clock. |
14:47.10 | drray | yay #0 |
14:47.10 | drray | er |
14:47.21 | eKo1 | OK. If I want the span to supply the clock, then I should use 0? |
14:47.44 | coppice | yep |
14:47.49 | coppice | but you don't |
14:47.55 | brettnem | vader--: if it gets a exten that matches NXXXXXX it will attempt a call out g1 on ZAP.. After the call it will end in congestion or if the dial failed it will indicate congestion |
14:47.55 | tzanger | don't think of it as supply and demand. think of it as "am I going to try and sync to the far end of this span or not?" 1 = yes I will try to sync ot it, 0 = nope, I will do what I want |
14:48.44 | eKo1 | Because I plan to hook up a dialin server to one of the ports on my quad E1 card and I need to have it (the card) act as the clock source. |
14:48.54 | coppice | think "am I the centre of the universe" then calm your ego down and try that again. if the answer is "no", then don't use 0 |
14:49.45 | coppice | eKo1: then use 0 for that port, and make the dialin server slave to you |
14:49.58 | eKo1 | OK. Thanks. |
14:51.26 | *** join/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com) |
14:52.53 | frk2 | Agree with coppice. If you put 0 there you would get a lot of HDLC frame errors |
14:53.05 | frk2 | and dudes would complain of random call drops |
14:53.16 | eKo1 | OK. I made the changes and I rebooted the server. |
14:53.32 | frk2 | dont reboot man.. its Linux not windows :) |
14:54.10 | eKo1 | I know but I do it just in case. |
14:54.32 | frk2 | guys... i BADLY neeed random suggestions on my phone lockup issues. ANYTHING. These grandstream/Cisco lockup issues happen only at a particular client |
14:54.38 | eKo1 | The notice is gone. Great. |
14:54.57 | eKo1 | Define lockup. |
14:55.15 | frk2 | Lockup = 1) phone ceases to respond... need to power cycle |
14:55.37 | frk2 | OR in the grandstreams case sometimes the screen goes blank, no voice but if I dial the other phone rings (no voice again) |
14:55.45 | [TK]D-Fender | frk2 : I've heard the 8.0 family of Cisco SIP firmware is flakey and people have reverted back.... as for GS, I hear everything is flakey and people just send THEM back :) |
14:56.07 | frk2 | TKD- I cannot seem to find the 7.x SIP firmware |
14:56.13 | frk2 | for cisco's |
14:56.25 | drray | I'd never deploy grandstreams but my budgetone has been "serviceable" |
14:57.00 | frk2 | whats a good mid range phone then? |
14:57.19 | frk2 | Polycom/Aastra? |
14:57.22 | drray | I like the cisco 79x0's but that is not an asnwer |
14:57.23 | drray | for you |
14:57.36 | eipi | how i can configure my iaxy to register a named server? |
14:57.38 | frk2 | can somebody tell me where to get the older firmware please? |
14:57.38 | eKo1 | frk2: I have that happen to me with Grandstream phones and Sipura ATAs. |
14:57.41 | eKo1 | No biggy. |
14:57.56 | frk2 | eK01 - the locking up part? |
14:57.58 | drray | cisco.com has teh older firmware if you have a smartnet contract |
14:58.10 | frk2 | I dont have a smartnet contract :) |
14:58.10 | eKo1 | Yes, the don't work and have to be power-cycled. |
14:58.30 | eKo1 | s/the don't work/they stop working/ |
14:58.53 | eKo1 | :) |
14:59.00 | frk2 | Damn... hanging is total loss. |
14:59.12 | frk2 | My problem with Aastra is they dont make 220v power adaptors!!!! |
14:59.22 | frk2 | I guess I need to 802.3af' them |
14:59.40 | eKo1 | uh oh, the HDLC notice is back again. |
14:59.42 | eKo1 | rats |
15:00.57 | frk2 | whats the exact notifce? |
15:01.12 | eKo1 | pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 2 |
15:01.20 | [TK]D-Fender | frk2 : Polycom is quality stuff at a great price point. |
15:01.40 | frk2 | yeah man... will the 300/301 work right out of the box with SIP support? |
15:02.14 | [TK]D-Fender | frk2 : yup |
15:02.23 | [TK]D-Fender | frk, though I'd recommend the IP430 in its place. |
15:03.04 | dlynes_home | eKo1: yeah...i've been getting the same notice now for some time...it shows up every once in a while |
15:03.25 | dlynes_home | eKo1: it seems to be related to missing an interrupt here and there |
15:03.31 | frk2 | there is no IP 430 TKD |
15:03.35 | frk2 | 301/501 |
15:03.40 | frk2 | shit-- 301 is now $129 at telephonyware!!! |
15:03.43 | frk2 | im getting it |
15:03.44 | dlynes_home | frk2: yeah, there is |
15:03.45 | [TK]D-Fender | frk2 : http://www.polycom.com/products_services/0,1443,pw-34-182-15672,00.html |
15:03.46 | eKo1 | but zttool reports no irq misses |
15:03.49 | dlynes_home | frk2: check polycom's website |
15:03.57 | [TK]D-Fender | frk2 : IP 301 = $115 at Atacomm.... |
15:04.15 | dlynes_home | eKo1: Does your patlooptest show a pass on all of your spans? |
15:04.47 | eKo1 | patlooptest? Is that when I choose loop in zttool on the span? |
15:04.49 | *** join/#asterisk ToyMan (n=stuq@74-32-67-126.dsl1.mdl.ny.frontiernet.net) |
15:05.10 | dlynes_home | eKo1: no, that's when you type ./patlooptest /dev/zap/1 |
15:05.43 | dlynes_home | eKo1: make sure you use the clear option in zaptel.conf for whatever span you're testing |
15:05.54 | frk2 | why the hell dont they make 220v power supppliessssssssssssssssssss |
15:05.57 | [TK]D-Fender | frk2 : But do keep in mind you need to get the 301 with its special PoE cable which is a little bulky. IP 430 has a lot of perks that make up the difference. |
15:06.14 | [TK]D-Fender | frk2 : Go PoE with IP430's...... |
15:06.30 | eKo1 | dlynes_home: I don't have that tool. |
15:06.41 | dlynes_home | eKo1: yes you do...you just haven't compiled it |
15:07.03 | dlynes_home | eKo1: go into zaptel source directory and type make patlooptest |
15:07.27 | eKo1 | OK. |
15:07.38 | }btorch{ | do I need to recompile asterisk to have it run as a regular user ? or can I just change permissions and runas the user that I want it to run as |
15:07.52 | frk2 | hmm.. see this is the kind of stuff i dont understand.. GXP 2000 is the best seller at atacomm!! :) |
15:07.53 | dlynes_home | eKo1: then put a t1 loop connector into the span you want to test |
15:08.29 | eKo1 | t1 loop connector? |
15:08.34 | dlynes_home | eKo1: yeah |
15:09.16 | [TK]D-Fender | frk2 : The sell a lot of CRAP... doesn't make it GOOD :) |
15:09.20 | *** join/#asterisk _4d4m_ (n=adam@62.69.102.99) |
15:09.25 | Hmmhesays | blargh |
15:09.34 | Hmmhesays | where did buildroot hide the kernel source |
15:09.48 | *** join/#asterisk ivanfm (n=ivanfm@c9068840.virtua.com.br) |
15:09.55 | [TK]D-Fender | frk2 : and what kind of marketing genius would label it like "Yeah it sucks, but look at the price!" |
15:10.00 | feld_ | wheee! i have asterisk working now with voicemail. so neat. :) now to get outside dialing working after updating asterisk.... that will be my afternoon goal. |
15:10.15 | [TK]D-Fender | feld : AFTERNOON? eek |
15:10.20 | eKo1 | I don't have one of those. |
15:10.26 | dlynes_home | eKo1: http://66.102.7.104/search?q=cache:9zhgfJEhcR0J:www.effeng.com/vtc/jseries/T1_Hardloop_Pinout.pdf+what+is+a+t1+loop+connector%3F&hl=en&gl=ca&ct=clnk&cd=1 |
15:10.41 | dlynes_home | eKo1: make sure you use the A standard (1, 2, 4, and 5) |
15:10.44 | frk2 | Dude this is my problem: |
15:10.49 | frk2 | The client doesnt have POE |
15:11.09 | eKo1 | dlynes_home: I have one of those already. |
15:11.13 | frk2 | If they have to install a POE switch just for this- its too much.. |
15:11.20 | dlynes_home | eKo1: i thought you said you didn't have one? |
15:11.22 | frk2 | what input voltage does the Polycom 301 have? |
15:11.28 | eKo1 | I didn't know you called it that. |
15:11.35 | dlynes_home | lol |
15:11.37 | eKo1 | I just call it a self looped e1 cable. |
15:11.37 | [TK]D-Fender | frk2 : Hrm..... ok, I know Polycom does have international power standard bricks... just not sure on the part # |
15:12.03 | dlynes_home | eKo1: yeah...e1 might be the b standard...I don't know |
15:12.47 | dlynes_home | anyways...i've gotta run |
15:13.14 | dlynes_home | eKo1: if you have any further problems getting the patlooptest set up, you might be able to find someone from digium that can help you with it |
15:13.33 | *** join/#asterisk wunderkin (i=kev@69.26.192.234) |
15:13.37 | frk2 | do you guys know of a good online / offline VOIP retail store location in Asia/Middle EAsy? |
15:13.40 | frk2 | East |
15:13.43 | dlynes_home | eKo1: or you can wait until i'm back in the office |
15:13.52 | dlynes_home | I'll be in the office in about 3 hours or so |
15:14.18 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
15:14.38 | *** join/#asterisk nahirean (n=nahirean@unaffiliated/nahirean) |
15:14.49 | s-ndh-c | hehe |
15:15.26 | *** join/#asterisk rvhi (n=rv@66.175.65.89) |
15:16.23 | eKo1 | dlynes_home: thanks |
15:17.32 | feld_ | [TK]D-Fender, shouldnt be that hard. I have the hardware and connections. I have some account information sitting in my mailbox. Just have to plug it all in I guess :) |
15:18.53 | eipi | anyone: how i can configure my iaxy to register a named server? |
15:19.06 | [TK]D-Fender | feld : Meant it should take MINUTES. |
15:19.24 | russellb | eipi: you can't |
15:19.48 | [TK]D-Fender | russellb : Still no DNS on that thing huh? |
15:20.05 | russellb | negative |
15:20.19 | russellb | who needs dns, pfft ... |
15:20.28 | [TK]D-Fender | russellb : If you want it to sell it needs a fair amount more work on the interface & functionality side... |
15:21.00 | [TK]D-Fender | russellb : Its a hard sell against ATA's like Linksys & co |
15:21.09 | feld_ | [TK]D-Fender, yeah but I have more than just this going on. I support a ton of businesses and their networks too. |
15:21.17 | russellb | gtkiaxyprov, man |
15:21.20 | russellb | :D |
15:21.49 | [TK]D-Fender | feld : Ok, *15 minutes* |
15:21.50 | eipi | :D |
15:22.08 | [TK]D-Fender | russellb : And still no DNS :) |
15:22.13 | frk2 | does anybody know the input voltage for Polycom 301/501? |
15:22.18 | feld_ | [TK]D-Fender, i still have to go power the damn thing down open it up and find out which are FXO and FXS lol |
15:22.19 | frk2 | is it 12v or 48V |
15:23.22 | *** join/#asterisk mr_horsepower (n=igor@82.102.1.42) |
15:23.27 | mr_horsepower | hi |
15:23.46 | [TK]D-Fender | frk2 : not sure.... I can tell you in about 10 hours..... |
15:24.20 | *** join/#asterisk vechers-away (n=svecher@64.61.117.139) |
15:26.18 | feld_ | anyone here use * with video? I just got a requst to find out what it takes.... |
15:28.36 | key2 | feld: I do |
15:28.39 | key2 | h263 |
15:29.06 | feld_ | what do your clients use for phones? |
15:29.22 | mut | damn i shoulda bought a house a long time ago |
15:29.31 | mut | combine car/house insurance |
15:29.40 | key2 | eyebeam |
15:29.46 | mut | brought my car ins payment down $80 |
15:29.57 | mut | and home ins pmt is only $40s |
15:30.03 | feld_ | nice |
15:30.33 | mut | ya, that freakin awesome, now i can afford to put that water softener in it |
15:30.36 | mut | :P |
15:30.39 | feld_ | my car insurance is through the roof. yay for getting screwed over. |
15:31.01 | mut | i pay $223/mo right now |
15:31.08 | mut | will be $152 |
15:31.45 | mut | year ago i was paying $293/mo |
15:32.06 | frk2 | TKd- thanks man- you are always a great help |
15:32.54 | [TK]D-Fender | frk2 : That sarcasm for the time its going to take me to get home? |
15:32.58 | feld_ | i honestly dont know what mine is atm. i think its near a grand a month =( i just graduated college, ins being payed by my parents until it comes down in january. so gay. accident which wasnt my fault and getting a ticket for sometihng i never did within like 2 weeks really screwed me over. that was 2 1/2 years ago. i hate this country :( |
15:33.09 | frk2 | TKD- no no dude... im serious.. you always help me out :) |
15:33.19 | Nugget | you get a discount for being gay? cool. |
15:33.28 | [TK]D-Fender | frk2 : Ok, can't be sure sometimes.. esp after a 10hour warning like that ;) np |
15:33.34 | frk2 | haha |
15:33.39 | mut | Nugget: well yea, it's being encouraged in society anymore... |
15:33.56 | frk2 | if you could tell me where to get the Cisco 7.x firmware from somewhere.. without the freaking contract, it would be aewsome |
15:34.01 | techman97_andy | gay discounts? wtf? that's what I get for minimizing this channel |
15:34.03 | frk2 | are resellers supposed to have this contract? |
15:34.13 | feld_ | rofl that reminds me the guy on the radio did the Sports section like he was gay. it was hilarious. |
15:34.25 | *** join/#asterisk Symm (n=s@198.87.2.15) |
15:34.33 | Symm | werd |
15:34.48 | fourcheeze | how does someone do a sports section like they're gay? |
15:34.52 | Symm | asterisk is being sold here in washington DC for beaucoup bucks |
15:35.01 | fourcheeze | do you mean "Camp" ? |
15:35.10 | kdz13 | fourcheeze: "that guy is so hot in his shorts!!!!" |
15:35.11 | Symm | its called, playing on the population's ignorance |
15:35.15 | Symm | hey no fag talk |
15:35.32 | fourcheeze | hmm |
15:35.45 | Symm | this is for visionaries only |
15:35.58 | *** join/#asterisk burizaa (n=freeee@cm66.omega101.maxonline.com.sg) |
15:35.59 | Symm | prophetic talk only please |
15:36.03 | fourcheeze | so a straight guy doing the sports section on a womens sport would be commenting on their bodies too? |
15:36.26 | [TK]D-Fender | frk2 : Lemme look |
15:36.39 | kdz13 | fourcheeze: I dunno, i never pay attention to sports |
15:36.43 | kdz13 | so i'm just guessing |
15:37.01 | kdz13 | but that's the only logical reason I can see why tenis playing women have to wear skirts |
15:37.19 | drray | tradition |
15:37.27 | drray | back when women were not allowed to wear pants |
15:37.37 | fourcheeze | hehe |
15:37.44 | fourcheeze | over hear "pants" are underwear |
15:37.46 | eKo1 | blame the british |
15:37.57 | mut | looks hot as hell on those courts anyway |
15:38.07 | drray | and they sell tennis outfits |
15:38.08 | mut | i wouldn't be in pants |
15:38.08 | fourcheeze | so wearing a skirt *because* you're not allowed to wear pants is quite an interesting thought |
15:38.26 | drray | well, we did not have berkas |
15:38.28 | drray | back in the day |
15:39.10 | *** join/#asterisk Overworked554 (n=Ken@atlantis.clearshout.com) |
15:39.45 | fourcheeze | I'm pretty sure I've seen women playing tennis in shorts |
15:40.06 | Hmmhesays | women with penii |
15:40.23 | Hmmhesays | that we generally refer to as men |
15:41.09 | mut | i think the proper term is hermaphrodite |
15:41.53 | fourcheeze | http://aeltc.wimbledon.org/en_GB/about/history/fashion.html |
15:42.30 | Vorondil | hi all, quick question: when using variables in a dial plan, the expression, "$[2${EXTEN:-2}CELL]" would end up as, say, say, "201CELL" (assuming the correct value of ${EXTEN}). what if "201CELL" is the name of a global variable? is there a way to get asterisk to stick the value of /that/ variable in there? |
15:42.50 | *** join/#asterisk SplasPood (n=jwb@206.252.198.101) |
15:42.51 | Vorondil | (sans the extra "say".. :-P ) |
15:44.13 | JackEstorm | ugh, has anyone had problems with Agents locking up the queue, and then leading to * hanging? |
15:44.19 | eKo1 | Vorondil: are you trying to make variables dynamically or something? |
15:44.29 | feld_ | JackEstorm, no but I feel your pain :( |
15:44.59 | Vorondil | eKo1: yeah, i suppose you could call it that. the global var is already defined though |
15:45.17 | [TK]D-Fender | Vorondil : ${2${EXTEN:-2}CELL} |
15:45.56 | Vorondil | [TK]D-Fender: so you don't need the sqare brackets for concatenation? |
15:46.01 | mosty | i have a TDM400P, when i pickup my handset, the * console says Starting simple switch on 'Zap/2-1' but as soon as i dial a number it says Hungup 'Zap/2-1' - it doesn't appear to be traversing my dial plan. what could be wrong? |
15:46.16 | JackEstorm | feld_: I can't belive that after all this time Agents is still buggy as hell, but all these problems say it is. |
15:46.22 | [TK]D-Fender | Vorondil : no. [] is for logical operations only. |
15:46.23 | mr_horsepower | gxp-2000 -> wonderfull :D |
15:46.48 | mut | how do yuo tell if an incoming call has no callerid? |
15:46.49 | mosty | my zapata.conf has context=voicemail , and extensions.conf has a single extension defined in the voicemail context |
15:46.50 | Vorondil | [TK]D-Fender: ahh, i gotcha. thanks ^_^ |
15:46.53 | Hmmhesays | <PROTECTED> |
15:47.02 | Hmmhesays | bah |
15:48.00 | mut | via the dialplan ofcourse.. |
15:48.03 | frk2 | did somebody mention gxp 2000? :) |
15:48.26 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
15:48.39 | mr_horsepower | frk2: wonderfull, everything works nice, i just dont know how do write the dial number with alpha-numeric numbers. |
15:48.54 | coppice | there is a press release about them using a TI DSP in the GXP2000 like its a new model. have they done a revamp of it? |
15:49.04 | frk2 | Horsepower- are you saying you have no issues with the GXP? |
15:49.22 | mr_horsepower | frk2: not even 1 |
15:49.27 | mr_horsepower | everything works nice. |
15:49.36 | frk2 | how many you have / how long you been using them / call load on the GXPs |
15:49.41 | frk2 | when did you buy them? |
15:49.59 | mr_horsepower | i'm just testing this one. |
15:50.02 | asterisk-dud | hello everyone |
15:50.15 | mut | anyone know if there is a way? |
15:50.21 | mut | cause callerid still comes in.. |
15:50.25 | mosty | no matter, i found the error |
15:50.27 | mut | but the call is passed as private |
15:50.30 | frk2 | ah |
15:50.31 | mut | so no callerid shows |
15:50.35 | frk2 | wait till you put it in production |
15:50.36 | frk2 | :) |
15:50.37 | mr_horsepower | we need good sip phones, and puting cisco away, i havent found any good phones until this one. |
15:50.38 | mut | on the user end.. |
15:50.51 | frk2 | its feature rich... just not reliable |
15:50.53 | drray | why would you get rid of cisco phones? |
15:51.02 | mr_horsepower | frk2: whats about the problem? |
15:51.04 | frk2 | if it didnt fuck up as much it would be an awesome phone |
15:51.10 | fourcheeze | mr_horsepower: have you tried the SPA-941 ? |
15:51.11 | *** join/#asterisk maik (n=maik@bfs.cs.uni-sb.de) |
15:51.18 | mr_horsepower | fourcheeze: no. |
15:51.20 | mr_horsepower | not yet |
15:51.23 | fourcheeze | works for me |
15:51.28 | Overworked554 | im having probs with my 7960. When an inbound call comes in and i pickup the receiver it hangs up on the caller. its running 8.3 |
15:51.31 | fourcheeze | as a basic but good phone |
15:51.46 | asterisk-dud | I would like to make a dialplan: _20XX that would take the extention dialed (I would have another variable with name equal to the exten, this variable would equal the channel to dial for that extension) |
15:51.51 | Overworked554 | other than that the ciscos work pretty well |
15:51.53 | mr_horsepower | fourcheeze: do you have sip url dialing? |
15:51.57 | mr_horsepower | with dots? |
15:52.03 | drray | my ciscos have been rock solid |
15:52.06 | }btorch{ | hey this is wierd , if I change my voicemail password throught the voicemailman options shouldn't that password be saved on my voicemail.conf file ? |
15:52.09 | fourcheeze | mr_horsepower: no idea |
15:52.10 | [TK]D-Fender | fourcheeze : SPA-941 is decent, but just not worth the money in North America.... |
15:52.11 | drray | but I am running 7.x something |
15:52.15 | asterisk-dud | i need the dialplan to take the exten and find the channel value for it and dial it |
15:52.37 | mr_horsepower | fourcheeze: well i need to know that. i will try to test that phone. how much you can find that sip phone? |
15:52.43 | asterisk-dud | any ideas? |
15:53.01 | fourcheeze | mr_horsepower: you can assign a sip url to a speed dial |
15:53.05 | fourcheeze | and dial it like that |
15:53.06 | mr_horsepower | the problem with sip phones it's, the good ones, are so much expensive. |
15:53.22 | drray | you get what you pay for |
15:53.31 | frk2 | yeah man.. why not get a polycom 501 for the same price as the 941 |
15:53.32 | drray | I"ve not had to dick with an expensive cisco phone |
15:53.47 | }btorch{ | it seems like everytime i reload app_voicemail.so or restart * the mailboxes are reset |
15:53.49 | mr_horsepower | fourcheeze: yes, can you try a sip url just like "foo.bar(at)domain.com" it works? |
15:53.56 | fourcheeze | frk2: how much is a polycom 501? |
15:54.08 | *** join/#asterisk mtaht3 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
15:54.15 | drray | if you value your time at $20/hr that 9 hours I spent screwing with the budgetone the time it decided to lose its mind |
15:54.18 | fourcheeze | mr_horsepower: I don't have one on my desk right now otherwise I would |
15:54.26 | [TK]D-Fender | fourcheeze : $170USD. IP430 is a great alternative @ $160 (so far) |
15:54.34 | fourcheeze | more expensive in the UK then |
15:54.36 | vader-- | can any of oyu guys explain to me what this does |
15:54.43 | mr_horsepower | drray: well, your hour it's cheap. but you are talking about 1 phone. |
15:54.45 | asterisk-dud | can i use variable arrays for asterisk dialplans |
15:54.47 | vader-- | exten => _NXXXXXX,1,Dial(ZAP/g1/${EXTEN:1}) |
15:54.47 | vader-- | exten => _NXXXXXX,2,Congestion |
15:54.47 | vader-- | exten => _NXXXXXX,102,Congestion |
15:54.53 | }btorch{ | am I configuring something wrong ? |
15:54.58 | fourcheeze | over here ip501 is about £180 |
15:55.07 | fourcheeze | ~ $350 |
15:55.14 | fourcheeze | usd |
15:55.15 | [TK]D-Fender | vader-- : It was already explained to you earlier |
15:55.21 | vader-- | i didn't see |
15:55.22 | fourcheeze | whereas the spa-941 is half that |
15:55.27 | [TK]D-Fender | fourcheeze : That royally sucks... |
15:55.39 | mr_horsepower | but what are the problems with gxp-2000? |
15:55.46 | [TK]D-Fender | fourcheeze : Yeah, thats why I said in North America there is no point to Sipura phones :) |
15:55.53 | fourcheeze | yeah |
15:56.02 | frk2 | well two problems ive faced so far: |
15:56.07 | frk2 | 1. It HANGS, randomly |
15:56.09 | Chotaire | hm, since the last kernel update on opensuse 10.0 I have problems running modprobing ztdummy (zaptel-1.2.4)... |
15:56.11 | Chotaire | # modprobe ztdummy |
15:56.11 | Chotaire | FATAL: Error inserting ztdummy (/lib/modules/2.6.13-15.10-default/misc/ztdummy.ko): Unknown symbol in module, or unknown parameter (see dmesg) |
15:56.17 | frk2 | 2. Screen goes blank and voice leaves |
15:56.23 | Chotaire | I never had this problem before.. anyone run across this shit yet? |
15:56.24 | [TK]D-Fender | vader-- : Scroll up |
15:56.25 | vader-- | oh ok i see now |
15:56.32 | mr_horsepower | who the hell uses suse to install a pbx? |
15:56.33 | mr_horsepower | :o |
15:56.33 | vader-- | someone said i should play the congestion sound? |
15:56.49 | sevard | mr_horsepower: correction, who uses suse |
15:56.50 | [TK]D-Fender | vader-- : Correct |
15:56.56 | Vorondil | [TK]D-Fender: hmm, using ${} doesn't seem to work. i get this (http://pastebin.com/762601) in the console |
15:57.00 | eKo1 | Chotaire: make clean and recompile |
15:57.02 | Vorondil | it ends up as a "" |
15:57.02 | drray | someone who uses suse for everything else in their shop |
15:57.07 | Chotaire | eKo1: that's what I did |
15:57.20 | Chotaire | like I do after every kernel update. |
15:57.30 | vader-- | is it just called conjection? |
15:57.32 | *** join/#asterisk aze (n=aze@ACayenne-101-1-12-4.w81-248.abo.wanadoo.fr) |
15:57.33 | vader-- | conjestion |
15:57.38 | mr_horsepower | frk2: the issues i have, was, in conference the phone crashed. firmware upgrade solved all my problems. |
15:57.51 | eKo1 | Chotaire: What kernel are you using? |
15:58.01 | [TK]D-Fender | Vorondil : There is another function to nest that... can't recall the name.... |
15:58.18 | frk2 | what firmware you using now? |
15:58.22 | frk2 | the new stable? |
15:58.45 | Chotaire | eKo1: found the bug... old zaptel was still running... |
15:58.47 | Chotaire | thanks anyway. |
15:59.03 | Chotaire | rmmod zaptel ; modprobe zaptel ; modprobe ztdummy ..fixed it. |
15:59.12 | Chotaire | duh ;) |
15:59.39 | sevard | It'd be sweet to have a soft phone sitting on a zapscan exten listening for fax and spitting them to pdf |
15:59.40 | sevard | ;) |
15:59.40 | sevard | evil |
16:00.13 | Vorondil | [TK]D-Fender: ah, okay. any idea the best place to look up such a function |
16:00.56 | mr_horsepower | frk2: Program-- 1.1.0.13 Bootloader-- 1.1.0.1 |
16:00.56 | [TK]D-Fender | Vorondil : WIKI time..... |
16:01.17 | frk2 | yup |
16:01.22 | Vorondil | [TK]D-Fender: hehe, kk. thanks much :) |
16:01.26 | frk2 | horsepower.. deploy the phone at a user and observe for 1 week |
16:01.45 | frk2 | Do not reboot, do not power cycle |
16:02.49 | }btorch{ | does * have a voicemail password memory I don't know about besides the voicemail.conf ? |
16:03.37 | [TK]D-Fender | }btorch{ : Nope. |
16:03.44 | sevard | So, is * 1.2.9 considered bleeding edge and 1.0.11 considered stable? |
16:04.21 | fourcheeze | 1.0.11 is considered old |
16:04.24 | [TK]D-Fender | sevard : no. 1.0.11 is for people who haven't done the big jump.. think of it like service packs to old versions... a waste if you ask me... |
16:04.25 | frk2 | no man.. 1.2.x is stable |
16:04.26 | fourcheeze | 1.2.9 is considered usable |
16:04.28 | frk2 | 1.0.x = legacy |
16:04.37 | frk2 | nice of asterisk dudes to support old legacy software |
16:04.41 | }btorch{ | [TK]D-Fender: then how come everytime I change the password the password in my #included file isn't changed and everytime I reload the app it thinks I'm new user |
16:05.00 | *** join/#asterisk jg (n=jg@1cc-dhcp-91.media.mit.edu) |
16:05.06 | eipi | anyone: how i can configure my iaxy to register a named server? can I? |
16:05.40 | zoa | sevard: what if i tell you that will happen soon ? :) |
16:05.44 | [TK]D-Fender | }btorch{ : PASTEBIN |
16:09.08 | vader-- | hmmm ok something is messed up in my zapata.conf |
16:09.21 | vader-- | when i dial ZAP/g1/${EXTEN:1} |
16:09.45 | vader-- | it rings the first available line in my zapata.conf instead of going out over the pri |
16:09.47 | vader-- | hehe |
16:11.41 | mosty | how do i create a voicemail mailbox? is there a script i can use? |
16:11.48 | swytch | quit |
16:11.48 | *** join/#asterisk timscott (n=a@d198-53-23-18.abhsia.telus.net) |
16:12.37 | Corydon-w | mosty: voicemailboxes are created automatically, as they are needed |
16:12.47 | *** part/#asterisk maik (n=maik@bfs.cs.uni-sb.de) |
16:13.11 | mosty | Corydon-w, i just tried to leave a message, and the asterisk console has errors about directories not existing |
16:13.35 | Corydon-w | Are you running as root? |
16:13.50 | MikeJ[Laptop] | do the directories exist ;) |
16:14.03 | *** join/#asterisk Niosop (n=Niosop@isd1.lvti.cc.nm.us) |
16:14.05 | mosty | Corydon-w, no it's running as the asterisk user |
16:14.09 | MikeJ[Laptop] | sounds like permissions... |
16:14.30 | Corydon-w | mosty: then you haven't allowed the asterisk user to own the /var/spool/asterisk directory |
16:14.33 | mr_horsepower | frk2: do you know any way to write alpha-numeric numbers in gxp-2000? |
16:14.54 | Niosop | Hello, anyone know of a patch that will force asterisk to send notify events to subscribed sip clients when max-calls has not yet been reached? Let me know if I'm not being clear. |
16:15.14 | mosty | Corydon-w, it is already owned by asterisk, and rwx for that user |
16:15.35 | Corydon-w | mosty: then something under that directory isn't |
16:16.22 | mosty | the console says it can't open this file for writing (no such file or dir): /var/spool/asterisk/voicemail/local/501/INBOX/msg0000.WAV, there is no voicemail dir in /var/spool/asterisk/ |
16:17.04 | feld_ | mosty, make one |
16:17.06 | feld_ | lol :P |
16:17.22 | feld_ | sounds like more than just that, though =( |
16:17.58 | mosty | after adding that dir, it let me leave a message. i'm surprised that this directory wasn't created when i installed asterisk (debian package) |
16:18.14 | feld_ | report-a-bug |
16:18.17 | feld_ | :) |
16:18.38 | mosty | it's a package from backports.org, i'm not sure where to report bugs |
16:18.46 | Corydon-w | Oh, you're not installing from source? |
16:18.55 | Corydon-w | There's your problem |
16:19.17 | Niosop | Problem: ext 100 subscribes to notify events (gxp-2k using blf), but asterisk doesn't send notify events if max-calls is greater than 1 and all the lines are not in use. Anyone know of a config option or patch that would force it to always send notifications? |
16:19.26 | Corydon-w | The source package works fine. It's all these packagers who screw up the system |
16:20.09 | *** join/#asterisk Splat (n=Splat@220-253-105-69.TAS.netspace.net.au) |
16:21.50 | mosty | it seems to be half working now. it creates subdirs are needed, writes out the files, but the files disappear when i hang up |
16:22.02 | feld_ | configure: error: C++ preprocessor "/lib/cpp" fails sanity check |
16:22.08 | feld_ | whats up with that ? any suggestions? |
16:22.16 | Niosop | you have glibc installed? |
16:22.18 | feld_ | happened b4 and after i updated my system. CentOS |
16:22.34 | feld_ | I would prefer gentoo but I never installed this damn box =/ |
16:23.04 | Niosop | yeah, gentoo is nice. Using A@H right now on CentOS though for testing. Production may end up being Gentoo if I have time. |
16:23.16 | sevard | Slackware my friend :) |
16:23.50 | feld_ | Niosop, keep a chroot up to date and periodically burn it to a dvd or something |
16:23.50 | feld_ | then u always have a gentoo install ready to deploy |
16:23.58 | feld_ | keep it like i686 or somethin |
16:24.20 | Niosop | shrug, going to be moving to xen VT server pretty soon hopefully, then I'll just have images ready to go. |
16:25.25 | feld_ | that would work too |
16:25.33 | feld_ | we run that VMWare ESX or whatever |
16:25.42 | feld_ | monitor all the VM's across multiple servers |
16:25.45 | feld_ | pretty cool setup |
16:25.50 | Niosop | nod. |
16:25.54 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
16:25.57 | feld_ | though it's based on windows which is sucky. |
16:26.04 | Niosop | yup |
16:26.12 | diLLec | oeh |
16:26.14 | feld_ | "yeah we can run linux" "but it's running on windows" "so what's the problem?" |
16:26.24 | Niosop | So nobody knows of any way to force notify events even if all available lines are not in use? |
16:27.02 | feld_ | sevard, used slack a few times. have a friend that really likes it. it's a nice quick system too. |
16:29.27 | *** join/#asterisk assert_true (n=Sunil@59.176.16.254) |
16:29.46 | mishehu | I'm looking in a recent zapata.conf file, and wondering why would zaptel need a jitter buffer |
16:29.58 | [TK]D-Fender | Yup... Slackware = trouble-free for me.... |
16:30.10 | Damin | field: Vmware ESX is NOT based on Windows. It runs a customized RedHat distribution for the Vmkernel. |
16:30.17 | sevard | feld_: I heart slax |
16:30.33 | Damin | field: Vmware "G"SX can run on Windows.. |
16:31.11 | Damin | filed: But that has been deprecated in favor of Vmware Server.. |
16:31.31 | feld_ | Damin, well whatever it is we're running it's based on Windows I believe |
16:31.40 | feld_ | I'll go check out the exact name later. I'm lazy. |
16:31.44 | feld_ | :P |
16:32.23 | Damin | feld: Well, Vmware Server can run on Linux or Windows.. |
16:32.23 | feld_ | omg how retarded guys |
16:32.34 | feld_ | error is because centos doesnt come with g++ |
16:32.46 | feld_ | how counter intuitive |
16:33.12 | [TK]D-Fender | feld : Did a MINIMAL install... I had a client that started the same way.... |
16:33.13 | Damin | feld: And if you are trying to run Asterisk on Vmware, you are in for a really crappy experience.. |
16:33.19 | feld_ | here's my next question: how did these guys get asterisk installed from source if g++ has never been on this system? |
16:33.33 | [TK]D-Fender | feld : I have to load at least a dozen packages to get everything he needed to get up and running... |
16:33.34 | feld_ | Damin, no i'm on a real server for Asterisk. |
16:34.07 | feld_ | I know it requires low latency high priority and nothing else eating its resources |
16:34.17 | Niosop | feld_, they compiled it on a different system, or removed the compiler toolchain after compiling? |
16:34.55 | feld_ | Niosop, no they arent smart enough |
16:35.10 | feld_ | for example: SSH is still open to passwords. |
16:35.21 | feld_ | they installed X |
16:35.32 | feld_ | these arent bright people we're talking about here lol |
16:35.45 | feld_ | at least they leave me alone though because they trust I know what I'm doing. |
16:36.05 | Hmmhesays | ok I have a stupid error now trying to compile an old version of zaptel |
16:36.08 | Hmmhesays | my paths are farked up |
16:37.04 | [TK]D-Fender | viler : Your PM's don't seem to be working.. you need to be registered on FreeNode for that. |
16:38.13 | *** join/#asterisk docE (n=docelmo@66.237.242.41.ptr.us.xo.net) |
16:38.30 | feld_ | [TK]D-Fender, u cant get into this chan without being registered |
16:38.43 | [TK]D-Fender | feld_ : Thats intermittant... |
16:38.53 | [TK]D-Fender | feld_ : Its deactivated occasionally. |
16:40.10 | feld_ | ic |
16:42.25 | *** topic/#asterisk by russellb -> Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- Asterisk 1.2.9.1 and 1.0.11.1 released, please upgrade immediately (June 6, 2006) -=- FreePBX/AMP/Asterisk@Home/Trixbox support in #freepbx |
16:42.38 | [TK]D-Fender | LOL! |
16:42.41 | russellb | :D |
16:42.42 | *** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com) |
16:42.45 | [TK]D-Fender | russellb : Already? |
16:42.48 | russellb | yes |
16:42.57 | feld_ | aww shit |
16:43.00 | russellb | there was a problem with the security fix :( |
16:43.02 | [TK]D-Fender | haha |
16:43.07 | feld_ | i just grabbed like 20 mins ago |
16:43.12 | [TK]D-Fender | russellb : the "fix" is in! |
16:43.13 | feld_ | am i up to date or not? ROFL |
16:43.21 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
16:43.26 | russellb | feld: not anymore :) |
16:43.29 | mitcheloc | *sigh* |
16:43.29 | [TK]D-Fender | feld_ : Dunno, ask in 20 mins ;) |
16:43.35 | feld_ | HAHAHA |
16:43.38 | mitcheloc | russellb: lol, is this in cvs? |
16:43.49 | sevard | how does one reload the manager.conf ? |
16:43.52 | [TK]D-Fender | mitcheloc : SVN & FTP I suspect |
16:43.52 | russellb | no more cvs |
16:43.55 | *** part/#asterisk mosty (i=mostynm@60-241-198-194.static.tpgi.com.au) |
16:43.59 | russellb | svn and ftp, yes |
16:44.14 | mitcheloc | russellb: haha, i mean svn, thats how long ago i updated ;) |
16:44.29 | russellb | feld_: you can actually just download a patch against the previous release |
16:44.33 | russellb | no need to download it all over again |
16:44.42 | russellb | that's why we release patch files with out releases :) |
16:44.51 | feld_ | russellb, lol ok thx |
16:44.56 | feld_ | i'll go do that :) |
16:45.01 | russellb | cool |
16:45.05 | [TK]D-Fender | russellb : How far back does the exploit go? |
16:45.14 | russellb | [TK]D-Fender: very far |
16:45.19 | mtaht3 | gah |
16:45.20 | [TK]D-Fender | russellb : O |
16:45.40 | mtaht3 | updating 15 servers now... |
16:45.42 | mitcheloc | russellb: is there a mailing list for security alerts? |
16:46.04 | russellb | mitcheloc: asterisk-announce gets all release announcements, which would include security alerts |
16:46.21 | mitcheloc | cool, i'll sign up on it then |
16:46.37 | [TK]D-Fender | I just troll in here..... works plenty fine.... |
16:46.53 | docE | so any other security issues other than the IAX that just showed up? |
16:47.54 | Hmmhesays | ok what is a *.a file ? |
16:48.09 | docE | a lib |
16:48.13 | docE | or library |
16:48.21 | Hmmhesays | does it matter what that library was compiled for? |
16:48.24 | docE | but it hasnt been compiled to be shared.. You have to link it. |
16:48.33 | docE | Its just a library |
16:48.41 | Hmmhesays | forgive the n00b question |
16:48.42 | Hmmhesays | is that a no? |
16:48.53 | russellb | what do you mean, what it was compiled for? |
16:48.56 | russellb | architecture? |
16:49.10 | Hmmhesays | yeah |
16:49.10 | docE | Depends on the library |
16:49.16 | russellb | yes, it absolutely does matter. |
16:49.32 | Hmmhesays | ok |
16:49.36 | sevard | how does one reload the manager.conf ? |
16:49.45 | vader-- | any of you guys know a good included sound with asterisk that would be good to play if someone tried dialing a 1900 number? |
16:49.45 | docE | stop now |
16:49.49 | docE | safe_asterisk |
16:49.59 | sevard | that's the only way? :| |
16:50.00 | mitcheloc | sevard: asterisk -vvvvr, reload |
16:50.21 | russellb | the reload CLI command will do it ... |
16:50.35 | mitcheloc | does anyone know if you can get stdoutput through the manager api? |
16:50.39 | docE | vader-- I can make you one that say's "DAMNIT STOP CALLING PORN! |
16:50.40 | docE | " |
16:50.46 | russellb | mitcheloc: no |
16:50.57 | mitcheloc | russellb: not even with the System command eh? =/ |
16:51.05 | kdz13 | how to turn off sip debug |
16:51.10 | docE | sip no debug |
16:51.17 | russellb | mitcheloc: what are you trying to do? |
16:51.22 | docE | good lord.. Does no one used the wiki anymore? |
16:51.29 | russellb | docE: i hope not |
16:51.44 | docE | Why? |
16:51.47 | mitcheloc | russellb: i want to get some extended functionality through the manager api,, without the requirement of putting a server piece on the asterisk machine to proxy through |
16:51.51 | sevard | docE: I use the wiki when I need to look up something.. but it's a horrible resource. |
16:52.00 | kdz13 | docE: thanks |
16:52.31 | docE | I found it quite informable when I needed it initially.. I dont use it much now cause of my experience level.. |
16:52.35 | docE | But the book is good too |
16:53.15 | docE | I had one until a guy I worked with gave it to some dude from South Africa.. last time I ever lend ANYTHING to anyone.. |
16:54.36 | *** join/#asterisk jsaunders (i=jsaunder@S01060060971c5817.va.shawcable.net) |
16:54.48 | mishehu | hmm... my pri provider called yesterday to check on why my pri was in lockout. yet other than echo and choppy echo, I'm not actually seeing any issues on the console. wouldn't a pri in lockout give an alarm on my t110p card? |
16:56.53 | eKo1 | Is there a way to configure the te401p so that some ports use T1 and some use E1? |
16:57.05 | eKo1 | Or do they all have to be either T1 or E1? |
16:57.38 | znoG | can't find the DND setup voip-info.org |
16:59.00 | jsharp | You can set them to T1/E1 on a per port basis |
17:01.43 | cybergypsy | what do I have to do to get calls to sip:user@mydomain.com to enter my asterisk ? |
17:01.56 | Hogie | If Im getting: No D-channels available! Using Primary on channel anyway 24! and it also says Primary D-Channel on span 1 down, is that a problem with my machine or the circuit? They are swearing that it is showing 0 errors on the switch side, but im not getting irq misses or anything else on our * box. This is on an install that's worked for 1+ years |
17:04.10 | Niosop | Problem: ext 100 subscribes to notify events for ext 101 (gxp-2k using blf), but asterisk doesn't send notify events if max-calls is defined for ext 101. Anyone know of a config option or patch that would force it to always send notifications? |
17:04.30 | *** join/#asterisk austinnichols102 (n=austinni@70.46.69.131) |
17:04.41 | blitzrage | Niosop: bugs.digium.com if it exists... else, you'll have to make it and submit it |
17:04.55 | [TK]D-Fender | blitzrage : ! ! ! |
17:04.56 | eKo1 | jsharp: Can I do that programatically or do I have to do it on the card physically |
17:05.46 | Niosop | blitzrage, was hoping not to have to :) Gonna take me hours or days to sift through chan_sip.c and figure out what's going on. |
17:07.27 | eKo1 | jsharp: OK, I found something on asteriskguru about this. Thanks. |
17:09.00 | Hmmhesays | ok what does this mean stdtime/libtime.a: could not read symbols: Archive has no index; run ranlib to add one |
17:11.02 | b00mer | What does this mean in my * logs? "-- Requested transfer capability: 0x00 - SPEECH" |
17:13.23 | *** join/#asterisk steve___ (n=steve@store-fw.porchlight.ca) |
17:14.22 | blitzrage | Niosop: sounds like you need to hire a consultant than |
17:14.26 | mut | how do you tell if a call is coming in with CID blocked? |
17:14.56 | Niosop | blitzrage, lol, naaa, there's a pickup patch for 1.2 that I can probably look through to give me an idea of what is involved :) |
17:15.17 | blitzrage | mut: CID isn't used to negotiate the call setup |
17:15.44 | mut | if i want to block calls with no callerid |
17:15.46 | mut | it is |
17:15.47 | docelmo | b00mer it has to do with POTS or ISDN connections |
17:15.53 | *** join/#asterisk noky (n=noky@200.69.211.18) |
17:15.54 | noky | hi |
17:16.09 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
17:16.19 | blitzrage | mut: what technology delivery? |
17:16.20 | feld_ | can anyone here decipher this build error junk? |
17:16.23 | feld_ | http://sh.nu/p/1925 |
17:16.27 | *** part/#asterisk Overworked554 (n=Ken@atlantis.clearshout.com) |
17:16.28 | mut | pri |
17:16.32 | mut | to sip |
17:16.35 | noky | my asterisk logs in /var/log/asterisk/full ... i have two question, the first is how can i change this path and filename to log in other place... and the second question is how can i desactivate the logs ..? |
17:16.45 | noky | russellb: hi |
17:16.57 | [TK]D-Fender | feld_ : ... |
17:17.00 | [TK]D-Fender | ~centosbug |
17:17.01 | jbot | methinks centosbug is a problem with the latest Centos kernel (4.2 and 4.3). To fix it, edit the file /usr/src/kernels/2.6.9-34.0.1.EL-i686/include/linux/spinlock.h and change 'rw_lock_t' on line 407 to 'rwlock_t'. This is part of the 'kernel-devel' package. Note that you may be using the 'smp' or 'hugemem' kernels. Change the filename to suit. |
17:17.06 | noky | [TK]D-Fender: hi |
17:17.26 | [TK]D-Fender | noky : y0 |
17:17.26 | feld_ | [TK]D-Fender, you my friend are my hero. |
17:17.42 | noky | feld_: is my hero too... |
17:19.13 | mut | blitzrage? |
17:19.26 | blitzrage | mut: hrmmm... might be a command in zapata.conf to block calls without CallerID.. but my brain is realyl tired, and I'm not 100% sure since I've not done that before |
17:19.39 | mut | well it's kind of a selective block |
17:19.46 | mut | so zapata wouldn't work |
17:19.50 | blitzrage | there is exten => 1234/5555,1,NoOp() format where 5555 would be callerID to match on |
17:19.54 | mut | i need to do it in the dial plan |
17:20.00 | mut | i also can't do that |
17:20.06 | mut | because i do get the callerid |
17:20.13 | mut | i's just being passed as a private call |
17:20.15 | mut | it's |
17:20.21 | blitzrage | is there not a channel variable that has that info? |
17:20.33 | mut | maybe |
17:20.35 | blitzrage | README.variables? I'm not really sure if there is one or not |
17:20.43 | blitzrage | might be worth a look anyways |
17:20.47 | mut | i just know of the callerid variable |
17:20.51 | mut | i'll check |
17:21.22 | b00mer | What does this mean in my * logs? "-- Requested transfer capability: 0x00 - SPEECH" |
17:21.43 | mut | ${CALLINGPRES} possibly? |
17:21.54 | mut | testies |
17:22.13 | blitzrage | mut: hrmmm... maybe? See what NoOp(${CALLINGPRES}) gives you... I'm kinda curious |
17:22.46 | blitzrage | I'd even try it here, but I'm in training and don't have Asterisk setup on this box |
17:23.46 | mut | well |
17:23.51 | mut | tried call 3 different ways |
17:24.00 | mut | <PROTECTED> |
17:24.00 | mut | <PROTECTED> |
17:24.00 | mut | <PROTECTED> |
17:24.04 | *** join/#asterisk Vahram (n=Noname@83.139.6.86) |
17:24.05 | blitzrage | heh |
17:24.15 | blitzrage | useful... |
17:24.16 | mut | dunno what #3 is |
17:24.24 | b00mer | ~pb |
17:24.25 | jbot | i heard pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
17:24.26 | blitzrage | what were the 3 different ways? |
17:24.36 | blitzrage | oh come on... 3 lines is not a big deal to paste |
17:24.43 | b00mer | 3 lines of junk |
17:24.48 | mut | first was dialing in via our internal system, pbx -> adtran -> box |
17:24.56 | blitzrage | junk? we're trying to debug something in asterisk |
17:25.08 | mut | seconds was picking up an outside pots landline then dialing in -> adtran -> box |
17:25.18 | mut | third was same outside pots line with *67 |
17:25.27 | blitzrage | hrmmm... |
17:25.32 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.234.37.Dial1.SanJose1.Level3.net) |
17:25.39 | blitzrage | wonder if the codes are listed in chan_zap... |
17:25.43 | *** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.234.37.Dial1.SanJose1.Level3.net) |
17:25.56 | blitzrage | let me checkout the code here and take a look -- let me know if you find it before me |
17:25.57 | mut | wiki has it |
17:25.58 | mut | http://voip-info.org/wiki/index.php?page=Asterisk+cmd+callingpres |
17:26.01 | *** join/#asterisk chaoscon (n=ph33r@smartserv/ceo/chaoscon) |
17:26.02 | blitzrage | well thast handy |
17:26.22 | mut | 0 i guess means nothing exists.. |
17:26.33 | blitzrage | makes sense |
17:29.01 | blitzrage | hrmmm... was kinda hoping there was a funciton for that which would return a string instead of numeric value |
17:29.51 | tzanger | blitzrage: macro it |
17:30.03 | blitzrage | mut: you could also verify by doing a SetCallerPres() and then loop the call back in and do a NoOp() to see what the value is |
17:30.29 | blitzrage | if you wanted to verify the codes |
17:30.51 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-2-21.red.bezeqint.net) |
17:30.53 | blitzrage | tzanger: I just meant a dialplan macro that would convert the Pres number to a string for you |
17:31.04 | blitzrage | errr... not macro - function :) |
17:31.16 | blitzrage | like... CALLERID(pres) or something |
17:31.21 | tzanger | bitch bitch bitch |
17:31.26 | blitzrage | tzanger: look who's talkin! |
17:31.31 | mut | laff |
17:31.38 | *** join/#asterisk ian_k (n=ian@gateway.digium.com) |
17:31.50 | blitzrage | tzanger: you're the bitch master! :) |
17:31.53 | tzanger | heh |
17:32.24 | blitzrage | I'm just trying to learn from the best |
17:32.39 | tzanger | you just keep that in mind, my friend... |
17:32.46 | blitzrage | oh... I will |
17:32.49 | blitzrage | and how! |
17:33.04 | mut | man i mosquito bit my on the inside of my hand yesterday |
17:33.30 | mut | and it's like bruised now, aparently cause it couldn't raise my skin cause it's callused it had to go deep |
17:33.50 | [TK]D-Fender | [13:31] <blitzrage> tzanger: you're the bitch master! :) <- Thigh-Master on "squeeze" ? ;) |
17:33.58 | *** part/#asterisk ian_k (n=ian@gateway.digium.com) |
17:34.46 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
17:37.10 | Vorondil | hi all, quick question: does anybody know if there's a dialplan function that will take the contents of one variable and use it as the name of another. roughly: var1=var2; var2=foo; function(var1) returns foo. |
17:37.47 | Vorondil | eval() doesn't seem to do it for me (unless i'm missing something) |
17:37.49 | Juggie | WHAT? |
17:38.23 | *** join/#asterisk dpryo (i=hn@donatello.nesland.net) |
17:38.28 | Vorondil | lol, does that not make sense? |
17:38.50 | Juggie | no, not really. |
17:39.36 | [TK]D-Fender | Vorondil : Figured it out. |
17:39.37 | Juggie | because your words and your psudo code are totally different. |
17:40.49 | mtaht3 | dang it - an hour ago I started rolling out asterisk-svn |
17:40.58 | mtaht3 | on a bunch of servers |
17:41.04 | [TK]D-Fender | Vorondil : ${EVAL(2${EXTEN:-2}CELL)} |
17:41.28 | blitzrage | Vorondil: almost sounds like you need to use _${VAR} (notice the _ ) |
17:41.43 | [TK]D-Fender | blitzrage : Nope... EVAL :) |
17:41.56 | blitzrage | pfffft |
17:41.57 | [TK]D-Fender | http://www.voip-info.org/wiki/index.php?page=Asterisk+func+eval |
17:42.23 | mtaht3 | is the corrected security update in svn head? |
17:42.28 | mtaht3 | (as of when) |
17:42.35 | blitzrage | [TK]D-Fender: I told you the other day to stop being so damn smart |
17:42.38 | mtaht3 | s/head/trunk |
17:42.41 | blitzrage | mtaht3: yah |
17:42.45 | Vorondil | [TK]D-Fender: well, i found eval(). it sound like it does *exactly* what i want, but i can't seem to get it to work |
17:42.53 | Vorondil | i must be missing something |
17:42.57 | mtaht3 | blitzrage - as of? |
17:43.19 | blitzrage | mtaht3: as of yesterday I believe |
17:43.34 | [TK]D-Fender | Vorondil : Pastebin your attempt |
17:43.36 | blitzrage | r32403 |
17:43.41 | blitzrage | which is the rev of 1.2.9 |
17:44.05 | Vorondil | [TK]D-Fender: alrighty, hold on |
17:45.24 | mtaht3 | blitzrage: thx |
17:45.55 | blitzrage | nada problemo |
17:50.02 | Vorondil | [TK]D-Fender: http://pastebin.com/762878 |
17:50.03 | *** join/#asterisk jgoo (n=e4b80e21@athe730f-2169.otenet.gr) |
17:50.16 | Vorondil | do i need to wrap another eval() around that? |
17:50.27 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
17:51.41 | *** join/#asterisk chapeaurouge (n=chapeaur@user-85-201-82-146.tvcablenet.be) |
17:51.44 | [TK]D-Fender | Vorondil : No, another set of ${} I think |
17:52.07 | Vorondil | (err, that should be "called 318 from 207 on line 13) |
17:52.09 | Vorondil | okay |
17:52.18 | blitzrage | yah.... needs to actually be a variable in the format ${VARIABLE} as opposed to just VARIABLE |
17:52.19 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
17:53.40 | Vorondil | bah, now i just get " -- Executing Dial("SIP/207-1beb", "IAX2/username@teliax/|60")" |
17:53.53 | Vorondil | so it's making an empty string |
17:53.54 | [TK]D-Fender | Vorondil : pastebin.... |
17:54.51 | blitzrage | Set(var2=foo) |
17:55.00 | blitzrage | Set(var1=${var2}) |
17:55.15 | blitzrage | ${EVAL(${VAR1})} is the format I believe |
17:55.28 | blitzrage | s/${VAR1}/${var1} |
17:55.31 | Vorondil | http://pastebin.com/762893 :) |
17:55.59 | *** join/#asterisk JINDAL (n=root@220.226.79.207) |
17:56.01 | [TK]D-Fender | Vorondil : ok, try nexting the eval |
17:56.16 | Vorondil | [TK]D-Fender: nexting? |
17:56.22 | blitzrage | can you show the 2 vars you're setting? |
17:56.45 | [TK]D-Fender | nesting* |
17:57.01 | [TK]D-Fender | blitzrage : he's trying to evaluate the variable to pick up..... |
17:57.37 | Vorondil | [TK]D-Fender: i gotcha |
17:57.38 | blitzrage | ? |
17:57.41 | blitzrage | I'm confused |
17:57.44 | *** join/#asterisk harlequin516 (n=sham@65.39.84.194) |
17:57.47 | blitzrage | moreso than usual |
17:58.06 | JINDAL | huy guys, am a total newbie for asterisk......... any suggestions where shd i start wid .......... i hav installed asterisk do i also need Zaptel |
17:58.06 | [TK]D-Fender | Vorondil : like http://pastebin.com/762903 |
17:58.07 | JINDAL | Libpri, Addons, Sounds |
17:58.25 | [TK]D-Fender | JINDAL : Depends what you intend on usings |
17:58.28 | blitzrage | JINDAL: www.asteriskdocs.org and click on "Read the book online" on the left hand side |
17:58.48 | harlequin516 | Are cellphones and SIP phones protocol compatible? I mean codec and protocol? |
17:59.04 | *** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
17:59.06 | blitzrage | [TK]D-Fender: I think you're missing a $ are you not? |
17:59.07 | JINDAL | okey... |
17:59.40 | blitzrage | in front of the inside EVAL |
18:00.02 | harlequin516 | Will public PSTN TElco's use e164.org enum? |
18:00.08 | blitzrage | doubt it |
18:00.10 | [TK]D-Fender | blitzrage : Stop being so damned smart ;) |
18:00.10 | harlequin516 | Do any now? |
18:00.22 | blitzrage | [TK]D-Fender: lol ... just good at finding syntax errors... |
18:00.44 | [TK]D-Fender | blitzrage : Yes... even *I* make them... I'm just better at displosing of the witnesses :D |
18:01.00 | blitzrage | [TK]D-Fender: lol -- YOU make errors? Does not compute. |
18:01.07 | Vorondil | [TK]D-Fender: http://pastebin.com/762907 it still doesn't get to the contents of the global. |
18:01.16 | tzanger | blitzrage: not much computes with you. :-p |
18:01.23 | blitzrage | Vorondil: can I see how you're setting the variables? |
18:01.27 | blitzrage | tzanger: its true |
18:01.35 | Vorondil | indeed, just a sec |
18:01.53 | [TK]D-Fender | Vorondil : I'm convinced you'll need to do it in 2 stages at least.... go play around with it for a while :) |
18:02.40 | JINDAL | [TK]D-Fender, plzz gimme very short info on Zaptel, Libpri, Addons, Sounds |
18:02.52 | harlequin516 | Anyone have IAX service provider that doesn't send DTMF properly? |
18:03.21 | Vorondil | [TK]D-Fender: here's how he globals are defined. |
18:03.42 | [TK]D-Fender | JINDAL : I asked you want you intended to DO with *. why would I waste time describing stuff you may never need? |
18:03.47 | *** join/#asterisk Cresl1n (n=matt@gateway.digium.com) |
18:03.54 | Vorondil | [TK]D-Fender: so i should try setting the stuff with EXTEN in it to a new var, then eval()ing the new var? |
18:04.00 | znoG | anyone know how to define a blank variable in AEL? |
18:04.03 | *** join/#asterisk vinkega_farmer (n=v_farmer@snoopy.xs4all.nl) |
18:04.09 | [TK]D-Fender | Vorondil : Just a though... use AstDB... much easier for what you're doing.... |
18:04.14 | znoG | foo=""; sets the variable as the actual "", I want it to be just BLANK. |
18:04.24 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
18:04.35 | blitzrage | znoG: foo= ? |
18:04.42 | blitzrage | and don't add ; as its not PHP :) |
18:05.05 | znoG | this is AEL |
18:05.08 | blitzrage | ahhh |
18:05.10 | znoG | in AEL there *is* ; |
18:05.21 | blitzrage | yah -- you didn't say it was AEL :) |
18:05.22 | JINDAL | [TK]D-Fender, okey my need is to setup a voip+ePBX system integrated wid regular PSTN analog/digital line |
18:05.30 | znoG | 15:05 < znoG> anyone know how to define a blank variable in AEL? |
18:05.34 | Vorondil | [TK]D-Fender: AstDB, eh? i'll look into that |
18:05.40 | blitzrage | znoG: oh -- I totally missed that line :) |
18:05.45 | harlequin516 | What's AEL? |
18:05.46 | znoG | foo=; didn't work btw. |
18:05.51 | blitzrage | ~ael |
18:05.52 | jbot | rumour has it, ael is Asterisk Extension Language - a dialplan language with 'c like' syntax? |
18:06.08 | harlequin516 | Oooh |
18:06.26 | blitzrage | znoG: hrmmm... good question -- might want to email murf and ask him... |
18:06.46 | *** join/#asterisk Arno[Slack] (n=root@66-163-12-60.ip.tor.radiant.net) |
18:06.49 | [TK]D-Fender | JINDAL : then you need ASterisk + Zaptel + Librpri |
18:07.17 | [TK]D-Fender | JINDAL : Addons gives support for MP3's for MoH, etc... useful, but not always necessary. |
18:07.51 | JINDAL | okey, thanks [TK]D-Fender |
18:08.22 | JINDAL | one more query will d pdf frm asteriskdocs.org describe them....... or i need to look somewhere else |
18:08.50 | [TK]D-Fender | JINDAL : Go read The Book for a while... |
18:08.52 | [TK]D-Fender | ~book |
18:08.54 | jbot | book is probably a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
18:09.05 | JINDAL | oks, thanks |
18:11.18 | *** join/#asterisk rene- (n=rene-@dsl-200-67-175-250.prod-empresarial.com.mx) |
18:12.01 | *** join/#asterisk lars-ut (n=lars-ut@70.103.228.158) |
18:12.08 | rene- | mmm |
18:12.19 | rene- | has anyone experience using RAMI? |
18:13.33 | rene- | and is it dangerous to connect to the manager console to get a sample of manager output in a running system, just reading events no sending and just one connection |
18:14.16 | harlequin516 | Hmm, OKay I looked at AEL, I'm not convinced of merits above and beyond extentions.conf. What's the big appeal. I was hoping for a true touring complete language.. It appears to just have some superficial syntactical similarity with respect to operators. Having read the hints/warnings, I am afraid of wasting days trying to perfect this hazzzardous art, |
18:14.48 | harlequin516 | Does AEL just create addenda to the standard dial plans? |
18:14.49 | [TK]D-Fender | harlequin516 : Its cleaner if you do a lot of loops, but adds nothing that you can't do with standard extensions.conf |
18:15.11 | [TK]D-Fender | harlequin516 : The AEL processor compiles it BACK to extensions.conf std dial-plan... |
18:15.20 | harlequin516 | okay |
18:15.20 | *** join/#asterisk kaz0358 (n=kaz@kazg5.telecom.ksu.edu) |
18:15.34 | [TK]D-Fender | harlequin516 : Basically.. its worthless |
18:15.41 | harlequin516 | No need to add complexity for me right now, I suppose. |
18:16.06 | harlequin516 | It didn't give me the impression of being clean/usable. |
18:16.10 | kaz0358 | out of curiosity, anyone know why asterisk was re-released again today? was the iax2 security vulnerability not in the 1.2.9 release like they said it was? |
18:16.31 | harlequin516 | THough I do see how it has appeal compared to extensions.conf |
18:16.45 | *** join/#asterisk r0d3nt|m (n=RatMan@foster.stonedcoder.org) |
18:17.36 | Katty | [TK]D-Fender: did i read somewhere that asterisk has a module for reciving a signal from an alarm console? |
18:17.58 | }btorch{ | hmm vmail.cgi doesn't really allow a user to change much of their setting , righ t? |
18:18.41 | *** join/#asterisk Chris_Stevenson (n=Mif`@209.172.67.146) |
18:19.00 | harlequin516 | If I forward a Zap Channel to Dial(IAX2/sham@myco.biz.com/1001), will the DTMF coming from the ZAP channel convey to IAX as IAX DTMF frames? |
18:19.32 | Chris_Stevenson | Hello, I am hoping someone in here might be able to help me get in contact with anyone at NuFone. Their support phone numbers appear to be disconnected, and Support/sales@ emails go unanswered |
18:20.12 | blitzrage | kaz0358: channels/chan_iax2.c: clean up yesterday's security fix to not |
18:20.13 | blitzrage | <PROTECTED> |
18:21.08 | blitzrage | and: * callerid.c: Bug 7268 - Callerid leaks memory on error |
18:21.16 | Chris_Stevenson | Is anyone else using nufone? |
18:21.27 | blitzrage | kaz0358: I just checked the ChangeLog file -- you should do the same ;) |
18:21.31 | harlequin516 | Any clues about DTMF conveyance, across disparate channel types? |
18:21.53 | blitzrage | I only looked because I didn't know a .1 was released :) |
18:21.55 | [TK]D-Fender | Katty : Can you elaborate on your need? |
18:23.11 | Katty | [TK]D-Fender: my boss just sent me a message about an alarm reciever thingy... i /think/ he meant alarmreceiver.conf which i'm looking at right now |
18:24.04 | Katty | [TK]D-Fender: all he said is that it recieved signals from an alarm panel in your house. |
18:27.01 | *** part/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
18:27.09 | feld_ | what setting plays the pre-recorded message when you are pushed to someone's voicemail box? |
18:27.09 | [TK]D-Fender | Katty : news to me... going to look now |
18:28.13 | Katty | [TK]D-Fender: okies. |
18:28.21 | Katty | [TK]D-Fender: also! http://www.voip-info.org/wiki/index.php?page=Asterisk+config+alarmreceiver.conf |
18:30.03 | [TK]D-Fender | Katty : Beat you there :) Yeah... could be useful for something I guess.... |
18:30.19 | [TK]D-Fender | Katty : boss wants notifications on alarms? |
18:30.32 | Katty | [TK]D-Fender: i don't comprehend how it works. that's my issue |
18:30.58 | Katty | [TK]D-Fender: right now we don't have an alarm control panel attached to a wall somewhere. |
18:31.14 | *** join/#asterisk philippel (n=p_lindhe@c-24-19-186-72.hsd1.wa.comcast.net) |
18:31.16 | [TK]D-Fender | Katty : It logs a specific proprietary alarm monitoring language and allows you to trigger apps off it. |
18:31.35 | [TK]D-Fender | Katty : Oh.. then this is useless to you :)) its for interfacing with things you don't have :) |
18:31.42 | Katty | [TK]D-Fender: and even if we did, i can't imagine what sort of thingy we'd have going because of it |
18:31.53 | Katty | [TK]D-Fender: yeah, but one of our clients might like it |
18:32.52 | [TK]D-Fender | Katty : And your "thingy" would have to speak that specific language as well... basically its warrantee-less partially usable code....*maybe* :) |
18:33.07 | *** join/#asterisk tdi (n=tdi@reykin.pozman.pl) |
18:33.39 | Katty | [TK]D-Fender: so only some security panels can 'interface'? |
18:34.04 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
18:35.52 | [TK]D-Fender | Katty : Yup... its a specific protocol... and security world is very proprietary... |
18:36.05 | }btorch{ | is it possible to configure asterisk to keep all user's preferences ov the voicemail system into mysql ? |
18:36.43 | *** join/#asterisk Bert- (n=bert@i05v-87-90-132-119.d4.club-internet.fr) |
18:36.49 | Bert- | hello there |
18:37.14 | Bert- | does someone knows a good MGCP softphone for Linux plz ? |
18:37.19 | Katty | [TK]D-Fender: connects via phone line or cat 5 line? |
18:37.41 | [TK]D-Fender | Katty : Analog. its an FSK based tech like modem/CID |
18:38.13 | AltnTab | is there any way to specify in zapata.conf what echo cancelerr to use: MARK2 for example |
18:38.14 | asterisk-dud | y can't i get my call transfer to work, zap channel, it does nothing when i dial *2 or even press flash |
18:38.31 | Bert- | ~softphone |
18:38.32 | jbot | something that should be drug out into the street and shot |
18:38.32 | [TK]D-Fender | AltnTab : No, it is chosen at compile time. |
18:38.41 | tzafrir_laptop | AltnTab, no, those are defined at build-time |
18:38.47 | tzafrir_laptop | (of zaptel, not of asterisk) |
18:38.52 | Bert- | erf |
18:38.52 | AltnTab | so i have just to uncomment it ? |
18:38.58 | *** join/#asterisk kink0 (n=k@62.37.205.161) |
18:38.58 | Bert- | ~MGCP |
18:39.00 | jbot | it has been said that mgcp is Media Gateway Control Protocol |
18:39.00 | AltnTab | if i uncomment all ? |
18:39.02 | kink0 | hi |
18:39.07 | *** join/#asterisk rvhi (n=rv@66.175.65.89) |
18:39.09 | AltnTab | [TK]D-Fender, |
18:39.38 | MatsK | I begin to get an idea who have writen jbot's answers :-) |
18:40.00 | kink0 | any idea why SS IXC -> Asterisk , the IXC CDR counts the total call duration for all calls and log CDR even if BUSY/Congestion is returned ? |
18:40.38 | AltnTab | tzafrir_laptop, hm, i have uncomment MARK2 and set echocancel=yes, echotrainning=yes and still have haevy echo problems |
18:40.42 | Dr-Linux | what's new in version 1.2.9 ? |
18:41.19 | tzafrir_laptop | a fix to a security issue with IAX |
18:41.37 | [TK]D-Fender | AltnTab : Zaptel EC isn't always so great... thats when you should consider getting an EC enabled card.... |
18:41.41 | AltnTab | tzafrir_laptop, i've recompiled all, zaptel, asterisk, run fxotune, load it, restart asterisk, calls ok in-out but ugly echo |
18:41.44 | Dr-Linux | tzafrir_laptop: ok thanks |
18:41.49 | [TK]D-Fender | Dr-Linux : and thats 1.2.9.1 BTW :) |
18:42.14 | tzafrir_laptop | AltnTab, did you set opermode to the right value? |
18:42.28 | tzafrir_laptop | (before messing with fxotune) |
18:42.42 | AltnTab | tzafrir_laptop, rxgain, txgain ? |
18:43.04 | tzafrir_laptop | opermode is a parameter of the kernel module wctdm |
18:43.17 | AltnTab | tzafrir_laptop, no, haven't mess with that |
18:43.38 | AltnTab | tzafrir_laptop, where should i read more ? |
18:43.43 | Dr-Linux | what's wrong with GSM , i'm getting this very frequantly >> Jun 6 11:08:54 WARNING[4600]: codec_gsm.c:194 gsmtolin_framein: Invalid GSM data |
18:44.10 | tzafrir_laptop | AltnTab, in what country are you? |
18:44.24 | AltnTab | tzafrir_laptop, bulgaria |
18:44.34 | AltnTab | tzafrir_laptop, eastern europe |
18:44.58 | *** part/#asterisk rene- (n=rene-@dsl-200-67-175-250.prod-empresarial.com.mx) |
18:45.00 | tzafrir_laptop | AltnTab, not everybody here are americans ... |
18:46.02 | tzafrir_laptop | you should set opermode=BULGARIA |
18:46.11 | tzafrir_laptop | I believe that this is case-sensitive |
18:46.35 | tzafrir_laptop | that is: |
18:47.03 | tzafrir_laptop | options opermode=BULGARIA |
18:47.04 | lars-ut | example of Asterisk TSP with integrated CRM? |
18:47.08 | tzafrir_laptop | options wctdm opermode=BULGARIA |
18:47.35 | AltnTab | tzafrir_laptop, ok, tnx |
18:47.40 | tzafrir_laptop | (use the latter) in /etc/modprobe.conf or /etc/modprobe.d/zaptel if you have that file |
18:47.49 | *** join/#asterisk chaoscon (n=ph33r@smartserv/ceo/chaoscon) |
18:48.42 | brettnem | hey anyone else having trouble compiling asterisk svn head? |
18:49.22 | feld_ | brettnem, u gettin errors? |
18:49.30 | Dr-Linux | [TK]D-Fender: any idea why i got this error on CLI almost for hundred of times? |
18:49.33 | Dr-Linux | what's wrong with GSM , i'm getting this very frequantly >> Jun 6 11:08:54 WARNING[4600]: codec_gsm.c:194 gsmtolin_framein: Invalid GSM data |
18:49.45 | brettnem | feld_ yes, a menuselect error regarding app_osplookup |
18:49.45 | [TK]D-Fender | Dr-Linux : nope. |
18:49.57 | brettnem | it's probably user error.... |
18:50.07 | feld_ | no idea, brettnem, but i hope u get an answer |
18:50.15 | brettnem | great |
18:50.16 | Dr-Linux | someone post the same error on list, but no one answered him |
18:50.16 | brettnem | thanks tho |
18:50.34 | techman97_andy | Dr-Linux - do you have any new SNOM Phones running? |
18:51.08 | Dr-Linux | techman97_andy: never use SNOM phone. i use SJphone, Cisco phones and eyeBeam |
18:51.30 | techman97_andy | Dr-Linux: When did the messages start happening? |
18:51.31 | brettnem | yeah, I compiled svn head a few days ago no problem. but today I tried again with today's svn head and it won't compile |
18:52.22 | brettnem | any ideas?? |
18:52.30 | MikeJ__ | brettnem, chat with russellb.... |
18:52.39 | MikeJ__ | menuselect is his baby |
18:52.45 | *** join/#asterisk chin1 (n=Administ@c-68-84-57-212.hsd1.nj.comcast.net) |
18:52.51 | techman97_andy | Dr-Linux: like # of days ago or whatever - I had a lot of codec warning like that when I was running the newest eyeBeam client |
18:52.53 | brettnem | russellb: you around? |
18:53.01 | chin1 | how come this ata isn't giving me a dial tone at all ? |
18:53.04 | brettnem | thanks mike |
18:53.13 | brettnem | MikeJ__: can I just disable it? |
18:53.14 | techman97_andy | you have to speak nice to the ata |
18:53.16 | techman97_andy | =P |
18:53.26 | chin1 | im being really really nice |
18:53.37 | techman97_andy | do you see the ata register in the CLI? |
18:53.42 | brettnem | ~seen russellb |
18:53.55 | jbot | russellb is currently on #asterisk (19h 26m 56s). Has said a total of 58 messages. Is idling for 2h 2m 26s, last said: 'docE: i hope not'. |
18:53.55 | MikeJ__ | dunno.. havn't really looked at menuselect... I think it reads in a config file |
18:53.55 | [TK]D-Fender | chin1 : And it would help if you told us something USEFUL about your scenario |
18:53.55 | MikeJ__ | brettnem, try #asterisk-dev. |
18:54.00 | noky | Jun 6 15:06:37 DEBUG[32567] db.c: Unable to find key '12' in family 'SIP/Registry' |
18:54.03 | noky | what is that ? |
18:54.03 | blitzrage | menuselect allows you to tell Asterisk which modules to compile |
18:54.06 | brettnem | MikeJ__: thanks |
18:54.13 | Dr-Linux | techman97_andy: i'm running eyeBeam client since 5 months, never get this kind of warning |
18:54.14 | chin1 | techman97_andy: does the ata have to be registered to the server for it to give a dial tone ? |
18:54.23 | noky | i have a gateway sip trying to register with a username 12 and password 12 too... |
18:54.26 | noky | what is happened? |
18:54.27 | blitzrage | noky: means that the key '12' doesn't exist in the family SIP/Registry in the AstDB |
18:54.32 | MikeJ__ | blitzrage, yeah.. he is actually getting an error in it so it fails to compile |
18:54.36 | chin1 | [TK]D-Fender: im just setting up a linksys pap2 |
18:54.54 | blitzrage | MikeJ__: oh yah? where is the error? (sorry, I missed it the first time around) |
18:54.55 | [TK]D-Fender | chin1 : Yes, it will only give you dial-tone if it is registered to a server. |
18:55.13 | techman97_andy | I'm back |
18:55.16 | noky | it exists. :S |
18:55.25 | blitzrage | CunningPike: upgrade minor, or upgrade major? (1.0 -> 1.2 for example) |
18:55.25 | [TK]D-Fender | CunningPike : Slightly less so if you have to manually merge SpanDSP, but not a big deal. |
18:55.27 | techman97_andy | chin1: Fender is correct on that |
18:55.32 | chin1 | well im looking at my status screen adn it says that my "Registration State: Online" |
18:55.57 | [TK]D-Fender | chin1 : Registered to WHERE is the question.... |
18:55.59 | techman97_andy | Dr-Linux: That's just my experience with weird codec warnings like that - something new that hit the network that * doesn't know how to deal with properly |
18:56.12 | chin1 | [TK]D-Fender: i have my sip login to my server on the internet |
18:56.21 | [TK]D-Fender | chin1 : to a "sip show peers" and pastbin it. |
18:56.22 | [TK]D-Fender | ~pb |
18:56.24 | jbot | [pb] a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
18:56.24 | techman97_andy | assuming it's a SIP ATA - do a sip show peers |
18:56.32 | techman97_andy | hehehe - Fender, I'll let you take that one |
18:56.33 | techman97_andy | your |
18:56.37 | techman97_andy | you're typing faster than I |
18:57.50 | noky | blitzrage: |
18:57.57 | CunningPike | blitzrage: Very minor: 1.2.8 -> 1.2.9 -> 1.2.9.1 in the space of an hour :D |
18:57.59 | noky | i have realtime with mysql |
18:58.07 | Dr-Linux | techman97_andy: hhmm.. my understanding is that, it due to when an extension try to use different codecs. |
18:58.16 | blitzrage | CunningPike: doesn't upgrade easily? Seems to just need to do make install -> restart now |
18:58.18 | noky | it exists... this messages appears from a minutes ago |
18:58.30 | blitzrage | noky: realtime eh? sorry... don't use it... |
18:58.48 | CunningPike | blitzrage: I was being genuine - it's a snap |
18:59.06 | blitzrage | CunningPike: oh... I just saw a :\ and assumed you were sarcastic :D |
18:59.09 | chin1 | [TK]D-Fender: i knwo what your gonna say it says "NAT: NO" for methods http://pastebin.com/763074 |
18:59.14 | blitzrage | CunningPike: in which case I'd agree with you :) |
18:59.25 | blitzrage | took me only 10 mins to upgrade 3 servers last night |
18:59.25 | techman97_andy | Dr-Linux: that could very easily be the case as well, but my experience (just sharing it to maybe trigger a thought for someone else) was that my eyeBeam client was using GSM only for about 3 weeks and everything was OK until I went to eyeBeam 3...then it went nuts. Maybe it's related, maybe not. |
18:59.32 | CunningPike | blitzrage: It's great |
18:59.38 | chin1 | the netmask seems messed up too ... |
18:59.48 | noky | blitzrage: why ? |
18:59.55 | [TK]D-Fender | chin1 : And that's your ATA? |
19:00.06 | chin1 | [TK]D-Fender: yes thats the ata connected |
19:00.16 | blitzrage | noky: why? because RT adds unneeded complexity in my opinion |
19:00.24 | CunningPike | Tada! Connected to Asterisk 1.2.9.1 currently running on dogmatix |
19:00.29 | [TK]D-Fender | chin1 : Keep in mind the PAP2 is a DUAL port.... I only see 1 peer registered.... maybe you plugged it into the wrong jack :) |
19:00.29 | blitzrage | ok ... going to pay attention to class |
19:00.44 | chin1 | [TK]D-Fender: no im on the right jack |
19:00.52 | techman97_andy | chin1: Try the other jack |
19:00.57 | chin1 | omg i wasn't! |
19:01.00 | chin1 | wtf.... |
19:01.01 | *** join/#asterisk __jkj (n=mail@adsl-69-150-161-180.dsl.lgvwtx.swbell.net) |
19:01.13 | chin1 | i sware i looked like 20 itmes! |
19:01.19 | russellb | brettnem: hey, i'm back |
19:01.21 | russellb | what's up |
19:01.21 | [TK]D-Fender | I geeeeeeeevve you FEEEEEEESHE! |
19:01.24 | chin1 | no more drugs for me |
19:01.25 | techman97_andy | it's always the most basic thing that f*cks you in the end |
19:01.36 | chin1 | anyway why does the mask say 255.255.255.255 and why does it say nat=n ? |
19:01.37 | [TK]D-Fender | chin1 : No, just BETTER drugs. |
19:01.40 | techman97_andy | like forgetting to wear pants to the store. |
19:01.45 | [TK]D-Fender | chin1 : Suivent, NEXT!!!!!!! |
19:01.50 | chin1 | what ? |
19:01.56 | brettnem | russellb: getting an error on today's svn head |
19:02.00 | chin1 | omg a dial tone!~!!! |
19:02.02 | [TK]D-Fender | chin1 : Bilingual queue call :) |
19:02.09 | __jkj | My new iAXY works inside my network but not outside. Any ideas? |
19:02.17 | brettnem | russellb: apps_osplookup |
19:02.18 | *** join/#asterisk themikester60 (n=mikey@cpe-72-181-92-164.houston.res.rr.com) |
19:02.21 | chin1 | so after i dial 600 do ihave to hit # or just allways wait 10 seconds lol |
19:02.22 | brettnem | in menuconfig |
19:02.29 | [TK]D-Fender | __jkj : What IP is it looking for? |
19:02.33 | chin1 | yep! |
19:02.38 | chin1 | this is so cool |
19:02.42 | *** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane) |
19:02.47 | noky | blitzrage: yes... could be... but i need realtime |
19:02.51 | [TK]D-Fender | chin1 : you can tweak your ATA dialplan to make certain things instant. |
19:03.02 | noky | blitzrage: i trying to start up an ivr with asterisk+realtime + webpage in php |
19:03.05 | __jkj | It is looking for my private ip inside the network and my public ip outside the network. |
19:03.09 | noky | an ivr configuration... |
19:03.09 | *** join/#asterisk watchy (n=watchy@70.238.57.237) |
19:03.17 | blitzrage | noky: yah... just not sure if that stuff that is normally in AstDB gets moved to the DB or not... |
19:03.21 | chin1 | [TK]D-Fender: ok all that stuff is experiment but im still concered why its saying nat=n |
19:03.59 | [TK]D-Fender | chin1 : because you don't have nat=yes for your peer entry. |
19:04.02 | chin1 | and my netmask is not all 255 |
19:04.29 | chin1 | i have nat settings on both my ata and on my sip.conf |
19:04.41 | [TK]D-Fender | chin1 : the netmask implies taht the IP specified is precise. |
19:04.50 | *** part/#asterisk Chris_Stevenson (n=Mif`@209.172.67.146) |
19:04.53 | [TK]D-Fender | chin1 : Pastebin it.... its not lying you know... |
19:05.14 | techman97_andy | chin1: if it works, quit screwing with it |
19:05.15 | techman97_andy | :D |
19:05.21 | chin1 | http://pastebin.com/763100 |
19:05.40 | chin1 | tahts my _sip_users.conf |
19:06.02 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
19:06.26 | feld_ | what does this mean? : Jun 6 15:05:58 ERROR[2889]: chan_sip.c:10988 handle_request_subscribe: Got SUBSCRIBE for extensions without hint. Please add hint to 2001 in context local |
19:06.28 | [TK]D-Fender | chin1 : Dunno.... hmmm.. well if it works I wouldn't worry |
19:06.44 | [TK]D-Fender | chin1 : do "sip show peer methods" |
19:06.57 | chin1 | what am i looking for ? |
19:07.16 | [TK]D-Fender | chin1 : pastebin :) |
19:07.23 | *** part/#asterisk austinnichols102 (n=austinni@70.46.69.131) |
19:07.54 | chin1 | it says NAT: Alwasy |
19:08.05 | [TK]D-Fender | chin1 : link please :) |
19:08.43 | chin1 | http://pastebin.com/763114 |
19:10.01 | chin1 | i dont undersatnd why it needs my internal ip |
19:10.08 | [TK]D-Fender | chin1 : Ok... I'm stumped, but it works... so thts enough :) |
19:10.19 | chin1 | i had the ata set to dmz before |
19:10.21 | chin1 | but i changed that |
19:10.32 | chin1 | maybe if i can de-register some how ? |
19:10.41 | chin1 | i tried pulling hte power didnt' change teh value |
19:11.22 | [TK]D-Fender | chin1 : ah whatever... it ain't broke.... be happy you're up and running |
19:11.37 | chin1 | [TK]D-Fender: hey this is a test for setting up a real system! |
19:11.40 | chin1 | i cant play no games... |
19:11.58 | chin1 | why do i allways see these musiconhold messages |
19:12.07 | feld_ | they want you to listen |
19:12.11 | chin1 | http://pastebin.com/763126 |
19:12.17 | chin1 | yea but im not even on the phone |
19:13.03 | [TK]D-Fender | chin1 : Means exactly what it says\ |
19:13.29 | [TK]D-Fender | chin1 : you've got stuff to clean up. |
19:13.43 | chin1 | yea its all default files from debian conf package |
19:13.55 | chin1 | or is that passed down from teh asterisk source ? |
19:14.08 | [TK]D-Fender | chin1 : you should switch to native MoH and not use MPG123, and then find some MP3's to use there and make sure to have compiled the asteriask-addons package |
19:14.23 | [TK]D-Fender | chin1 : SCREW DEBIAN PACKAGES. Use the Source Luke! |
19:14.34 | chin1 | [TK]D-Fender: i think debian has it all compiled allready :\ |
19:14.37 | chin1 | hey common its easy! |
19:14.46 | chin1 | ill setup slack with some built sources on spare time |
19:14.55 | kdz13 | chin1: as far as I can tell, debian's is quite old |
19:15.01 | chin1 | right now were paying like 260$ a month for this stupid pstn links |
19:15.56 | chin1 | im gonna get a polycom 501 for the secretary but these cordless phoens with the ata are fine for the rest of us were never int he office anyway |
19:16.04 | [TK]D-Fender | chin1 : I never said to not use Debian.. just compile * from source. |
19:16.45 | [TK]D-Fender | chin1 : Get her an IP601. That way she can handle more calls, and maybe expand to the attendant modules,. |
19:16.48 | chin1 | [TK]D-Fender: yes i know but im more of a slack guy i just used debian because A: they have precompiled binaries, B: my server in denmark is using debian so it was good to test |
19:17.10 | chin1 | 601 ? i think that might be it let me go look |
19:17.33 | noky | i think that is a fucking bug |
19:17.34 | noky | Jun 6 15:30:19 DEBUG[1242] db.c: Unable to find key '12' in family 'SIP/Registry' |
21:16.46 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
21:16.46 | *** topic/#asterisk is Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- Asterisk 1.2.9.1 and 1.0.11.1 released, please upgrade immediately (June 6, 2006) -=- FreePBX/AMP/Asterisk@Home/Trixbox support in #freepbx |
21:17.22 | *** join/#asterisk Beirdo (n=gjhurlbu@unaffiliated/beirdo) |
21:19.15 | *** join/#asterisk Chris1004 (n=chrislro@c-68-49-240-217.hsd1.md.comcast.net) |
21:19.28 | terrapen | anyone using ICD? |
21:19.41 | terrapen | i'm trying to find the README mentioned on the wiki |
21:21.09 | *** part/#asterisk chin1 (n=Administ@c-68-84-57-212.hsd1.nj.comcast.net) |
21:21.11 | Chris1004 | what dial plan will make outgoing calls using a T400 card. Im not sure what to put in the extensions.conf file |
21:21.56 | *** join/#asterisk C4T3l (n=robert@216.54.143.2) |
21:22.01 | C4T3l | hello all |
21:22.35 | C4T3l | anyone ever install a sangoma a102 before, I'm having trouble |
21:22.45 | noky | hi |
21:22.51 | noky | anybody test the sipp with asterisk ? |
21:22.59 | terrapen | the sipp? |
21:23.05 | noky | yes |
21:23.10 | terrapen | what the hell is the sipp |
21:23.22 | noky | is a tester for sip |
21:23.50 | Hmmhesays | http://www.rowetel.com/ucasterisk/ucasterisk.html |
21:23.50 | mercestes | *stares at C4T3l* |
21:23.51 | noky | sipp generate much calls with your parameters (call per seconds, duration of call, etc) |
21:23.56 | Hmmhesays | there is decent instrutions there |
21:24.00 | terrapen | interesting |
21:24.06 | noky | i don't know why my sipp say this: |
21:24.07 | noky | 2006-06-06 18:00:23: (1) No valid Call-ID: header in reply 'SIP/2.0 100 Trying |
21:24.10 | noky | :( |
21:24.15 | noky | any ide? |
21:24.17 | noky | idea? |
21:24.28 | noky | my asterisk answer OK, but the sipp quit whit this error... |
21:24.34 | C4T3l | i guess no one's ever done it?? |
21:25.13 | C4T3l | The installation of the wanpipe program dies before it can finish, no output or anything! |
21:25.42 | terrapen | no kidding |
21:25.55 | terrapen | or learn about strace |
21:27.07 | Juggie | noky, ask oej when hes online. |
21:28.21 | *** join/#asterisk chin1 (n=Administ@c-68-84-57-212.hsd1.nj.comcast.net) |
21:31.01 | chin1 | can i login via sip from multiple places at once ? |
21:33.54 | zoa | i tested sipp with asterisk before |
21:34.23 | Juggie | chin1, no asterisk does not support that. |
21:34.24 | noky | i found the error.. |
21:34.27 | Juggie | i wish it would, but it wont ;) |
21:34.32 | noky | was compactheaders = yes |
21:34.39 | noky | in sip.conf |
21:34.52 | Juggie | you can login if you like more then once |
21:34.59 | Juggie | but the latest phone to register will receive the calls |
21:35.08 | chin1 | oh |
21:35.16 | chin1 | spooky |
21:35.22 | Juggie | indeed. |
21:35.23 | mog_work | just give different ids |
21:35.29 | mog_work | but you can tie em together still |
21:35.37 | *** part/#asterisk Arno[Slack] (n=root@66-163-12-60.ip.tor.radiant.net) |
21:35.38 | Juggie | i had a good idea for that but no time to code it. |
21:35.40 | chin1 | my ata for some reason is not causing my cordless phone to ring |
21:35.40 | dlynes_office | ~seen mitcheloc |
21:35.44 | jbot | mitcheloc is currently on #asterisk, last said: 'russellb: i want to get some extended functionality through the manager api,, without the requirement of putting a server piece on the asterisk machine to proxy through'. |
21:35.54 | dlynes_office | mitcheloc: ummm...jbot's not on vacation |
21:35.58 | dlynes_office | mitcheloc: he just doesn't like you |
21:36.03 | Juggie | i would like to see SIP/sipphone be the last phone to register. |
21:36.05 | Juggie | but say |
21:36.19 | Juggie | SIP/sipphone[all] be all phones registered w/ that peer name |
21:36.28 | chin1 | the screen lights up but i dont get any ringing.. and yes the ringer is set to on |
21:36.55 | generalhan | ~dict Hmmhesays |
21:36.59 | generalhan | lol |
21:37.07 | Hmmhesays | heh |
21:37.13 | chin1 | dam i was gonna say someoen really answered that fast |
21:37.18 | dlynes_office | mitcheloc: however, it looks like he hasn't been logging peeps' logins and logouts and channel talk for a while |
21:37.19 | chin1 | lol |
21:37.48 | generalhan | dlynes_home: yeah he wasnt answering Hmmhesays a bit ago when he was asking for ~docs and ~thebook |
21:37.58 | dlynes_office | ~book |
21:38.00 | jbot | extra, extra, read all about it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
21:38.04 | file | he was gone you dingos :P |
21:38.04 | dlynes_office | That's why |
21:38.05 | Hmmhesays | i wasn't asking for either of those |
21:38.15 | dlynes_office | ~thebook |
21:38.21 | Hmmhesays | i had a cross compiling question |
21:38.21 | dlynes_office | There's no entry for '~thebook' |
21:38.34 | generalhan | Hmmhesays: no you werent ... i tried to scroll up and correct myself real fast but my scrollback is already past that ! lol |
21:38.45 | Hmmhesays | heh |
21:39.06 | generalhan | it was:: <ManxPower> ~docs |
21:39.20 | dlynes_office | yeah...you can only ask jbot about what he knows..not what he doesn't know |
21:39.28 | generalhan | yea but .... |
21:39.31 | generalhan | ~docs |
21:39.32 | jbot | somebody said docs was probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
21:39.40 | *** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
21:39.41 | dlynes_office | and your point? |
21:39.41 | generalhan | see i think he was on a smoke break ! lol |
21:39.58 | dlynes_office | <file> he was gone you dingos :P |
21:40.09 | dlynes_office | but i guess generalhan only sees what he wants to see :) |
21:40.10 | generalhan | was he ? |
21:40.16 | generalhan | damn straight |
21:40.18 | generalhan | lol |
21:40.27 | generalhan | why would i want to see what i dont want to see ?> |
21:41.11 | generalhan | HAHAHA it even says just a bit ago that jbot has joined #asterisk ! |
21:41.57 | generalhan | dlynes_home: ya know .. i have enough issues with my stupid 7960 firmware; without you slapping me |
21:42.03 | dlynes_office | lol |
21:42.13 | *** part/#asterisk C4T3l (n=robert@216.54.143.2) |
21:42.27 | dlynes_office | I can't believe cisco has the gall to charge that much for a phone with broken firmware |
21:42.31 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
21:42.34 | *** join/#asterisk Wowzers10 (n=pbaker@nnat-gw.adeptra.com) |
21:42.34 | generalhan | lol |
21:42.45 | *** join/#asterisk kjs3 (n=quux@c-24-98-110-80.hsd1.ga.comcast.net) |
21:42.53 | Wowzers10 | hello all, I just bought a Digium Wildcard TE410P, and noticed it was a 1 gen card - is it possible to upgrade firmware on these devices? |
21:43.00 | generalhan | dlynes_home: it sux cause eveytime i upgrade i think its all solved .. a week goes by then BAM back to swuare one |
21:43.00 | dlynes_office | but what's more unbelievable is people pay them that much for a phone with broken firmware |
21:43.30 | dlynes_office | Wowzers10: just bought it on ebay? |
21:43.35 | generalhan | dlynes_home: when i bought these phones we were using CM so they worked GREAT. then when i came over to * i didnt want to buy new phones so i started using them with the SIP firmware |
21:43.41 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
21:43.43 | Dr-Linux | dlynes_home |
21:44.02 | Dr-Linux | please check this this is really kililng me >> http://pastebin.com/763501 |
21:44.06 | dlynes_office | generalhan: ah |
21:44.33 | vader-- | any of you guys know a good included sound with asterisk that would be good to play if someone tried dialing a 1900 number? |
21:44.46 | vader-- | like don't dial this number pervert |
21:44.47 | vader-- | :) |
21:45.06 | dlynes_office | Dr-Linux: what's the two lines before the lines you pastebinned? |
21:45.08 | generalhan | hahaha |
21:45.08 | generalhan | yea |
21:45.09 | chin1 | i cant get my phone to ring! |
21:45.25 | generalhan | vader--: teletubbie-murder |
21:45.36 | dlynes_office | yeah...that'd be a good sound....1-900-I-CANT-GET-MY-PHONE-2-RING |
21:46.09 | generalhan | i made all my users record their name for the directory ... so when ever they do something incorrectly i put their recording into the teletubies recording and it works nicely ! lol |
21:46.16 | chin1 | Enable IP Dialing: |
21:46.17 | chin1 | what is that ? |
21:46.18 | kjs3 | vader: someone had a bunch of MP3s of 70s era pr0n movie sound tracks on their web site (Google, I suppose). Just play that in an endless loop. |
21:46.30 | Dr-Linux | dlynes_home: lemme show you |
21:46.39 | dlynes_office | chin1: something yoiu probably dont' want...especially if you don't want your customers to know they're using voip |
21:47.01 | chin1 | um |
21:47.07 | Dr-Linux | dlynes_home: |
21:47.07 | Dr-Linux | make -C /lib/modules/2.6.9-34.ELsmp/build SUBDIRS=/usr/src/zaptel-1.2.6 modules |
21:47.07 | Dr-Linux | make[1]: Entering directory `/usr/src/kernels/2.6.9-34.EL-smp-i686' |
21:47.07 | Dr-Linux | <PROTECTED> |
21:47.12 | chin1 | im trying to get my phone to ring the ata for some reason is not rigning it |
21:47.13 | Dr-Linux | is this gcc library problem |
21:47.23 | dlynes_office | chin1: what hardware? |
21:47.37 | dlynes_office | Dr-Linux: no, that's normal |
21:47.40 | chin1 | linksys pap2t-na and a 900mhz cordless |
21:47.46 | *** join/#asterisk AltnTab (n=ecs@nrjsoft13.networx-bg.com) |
21:48.04 | Dr-Linux | dlynes_home: what could be the problem? |
21:48.17 | Juggie | Dr-Linux, centos? |
21:48.21 | dlynes_office | chin1: the line on the linksys pap2 that you're plugging your cordless into has not registered yet |
21:48.42 | Dr-Linux | Juggie: RHEL AS 4 |
21:48.43 | *** join/#asterisk hagler (i=hagler@psychozoo.com) |
21:48.44 | chin1 | dlynes_office: yes it has im using it i see the screen light up and i can read the caller id but hte phone just doesn't ring |
21:48.49 | dlynes_office | Dr-Linux: i suspect you've got a fubar typedef somewhere |
21:48.56 | Juggie | Dr-Linux, |
21:48.59 | Juggie | ~centosbug |
21:49.02 | jbot | somebody said centosbug was a problem with the latest Centos kernel (4.2 and 4.3). To fix it, edit the file /usr/src/kernels/2.6.9-34.0.1.EL-i686/include/linux/spinlock.h and change 'rw_lock_t' on line 407 to 'rwlock_t'. This is part of the 'kernel-devel' package. Note that you may be using the 'smp' or 'hugemem' kernels. Change the filename to suit. |
21:49.23 | dlynes_office | Juggie: he's runnning 2.6.9-34.EL-smp-i686 |
21:49.26 | dlynes_office | heh |
21:49.29 | Juggie | yes |
21:49.30 | Juggie | so |
21:49.32 | Juggie | use your brain |
21:49.40 | hagler | so anybody know about an issue with spool calls where they fail when the called channel sends any sort of progress |
21:49.41 | dlynes_office | I was...that's why i was laughing :0 |
21:49.42 | Juggie | and replace the version in the path |
21:49.52 | hagler | Jun 6 14:39:44 NOTICE[15746]: pbx_spool.c:269 attempt_thread: Call failed to go through, reason 3 == Spawn extension (default, 571, 1) exited non-zero on 'Local/571@default-60cd,2' |
21:50.00 | dlynes_office | that centosbug snippet should have rhel added to it |
21:50.09 | Juggie | drlinux, nano /usr/src/kernels/2.6.9-34.EL-smp-i686/include/linux/spinlock.h |
21:50.17 | Juggie | and follow the instructinos a few lines bank. |
21:50.18 | russellb | jbot: rhelbug is aka centosbug |
21:50.19 | jbot | russellb: okay |
21:50.19 | Juggie | *back |
21:51.11 | Juggie | jbot: tell russell to put on pants |
21:51.19 | MikeJ__ | heh |
21:52.51 | chin1 | dlynes_office: you read that ? |
21:52.57 | Juggie | hah |
21:52.58 | Juggie | [17:51] <jbot> No, juggie, I won't. (target invalid?) |
21:53.19 | dlynes_office | chin1: so you can see the incoming call on the phone? |
21:53.19 | Juggie | jbot: tell russelb to put on pants |
21:53.37 | chin1 | dlynes_office: yes... it works fine i just cant hear a ring |
21:53.42 | dlynes_office | Juggie: two s's, two l's in russellb |
21:53.47 | chin1 | obviously a ata config right ? |
21:54.00 | dlynes_office | chin1: does the phone work in a normal analog jack? |
21:54.03 | chin1 | Caller Conn Polarity: ? |
21:54.03 | kink0 | MikeJ__ I read about woomera, but I have a doubt, basically why are you ussing woomera instead the h323 included in asterisk ? at least if you are not ussing it like endpoint or gatekeeper |
21:54.10 | chin1 | dlynes_office: yes i tried that |
21:54.38 | dlynes_office | chin1: i can't remember...is there a setting for ring voltage on those devices? |
21:54.51 | chin1 | on the ata or hte phone ? |
21:54.54 | chin1 | the phone is set to ring |
21:54.57 | dlynes_office | on the ata |
21:55.09 | dlynes_office | also, what brand is the phone? is it a siemens or a panasonic? |
21:55.17 | chin1 | Dist Ring Setting? |
21:55.23 | *** join/#asterisk MoutaPT (n=MoutaPT@85.139.196.14) |
21:55.24 | dlynes_office | ring voltage |
21:55.32 | MoutaPT | any one with BRI cards experience? |
21:55.37 | dlynes_office | it will only be called that, and nothing else |
21:55.46 | chin1 | its an atlinks i think i bought it at radio shack |
21:56.23 | *** join/#asterisk znoG (n=gs@99-211-126-200.fibertel.com.ar) |
21:56.29 | chin1 | wait |
21:56.34 | dlynes_office | ? |
21:56.35 | chin1 | im looking at a whole section on ring settings |
21:57.08 | chin1 | um |
21:57.16 | dlynes_office | the ring voltage if it's there will probably be on the bottom end of the settings for the line you're trying to use |
21:57.26 | chin1 | cfwd ring splash len |
21:57.32 | chin1 | vmwi ring splash len |
21:57.35 | dlynes_office | ring voltage |
21:57.36 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
21:57.39 | chin1 | lol |
21:57.42 | chin1 | no voltage |
21:57.47 | dlynes_office | not ring splash, not ring tone, not ring ..., just ring voltage |
21:57.57 | chin1 | nada |
21:58.08 | dlynes_office | chin1: check on the regional page as well |
21:58.16 | chin1 | what on the regional page ? |
21:58.27 | dlynes_office | chin1: for ring voltage |
21:59.04 | dlynes_office | if you don't see the regional page, it could be that you're logged in as user instead of admin |
21:59.36 | Symm | can you make free international calls with some kind of crazy hookup with asterisk? |
21:59.38 | Dr-Linux | back |
21:59.39 | dlynes_office | chin1: you won't see line 1 or line 2 if you're logged in as user, either |
21:59.42 | Dr-Linux | Juggie: what should i changeeeeee? |
21:59.49 | chin1 | dlynes_office: i foudn it ! |
21:59.57 | dlynes_office | Symm: yeah, if you know the ip address of the person you want to talk to |
22:00.00 | chin1 | i know that man |
22:00.03 | Symm | hmm |
22:00.08 | Symm | thanks, any other way? |
22:00.11 | chin1 | it says 70 |
22:00.14 | *** join/#asterisk fholmes (n=fholmes@rrcs-24-227-237-197.sw.biz.rr.com) |
22:00.20 | dlynes_office | Symm: own your own telco :) |
22:00.29 | Symm | oh i see, so you're saying, you avoid the pstn? |
22:00.42 | Dr-Linux | Juggie: can you tell me what should i change? |
22:00.53 | Symm | otherwise, you HAVE to go through a pstn and that costs money no matter waht |
22:00.58 | Symm | ? |
22:00.59 | dlynes_office | Symm: no, what i'm saying is that for the most part, using asterisk you don't avoid paying long distance, but you reduce the cost of your long distance |
22:01.01 | Symm | correct? |
22:01.06 | Symm | ok |
22:01.12 | dlynes_office | Symm: by avoiding pstn |
22:01.21 | dlynes_office | chin1: yeah...70's fine |
22:01.30 | dlynes_office | chin1: do you have a digital multimeter? |
22:01.30 | chin1 | dlynes_office: well thats not working |
22:01.40 | chin1 | no |
22:02.00 | chin1 | you think its the cord ? |
22:02.15 | dlynes_office | chin1: try taking the pap2 back to the store where you got it from then, and getting them to replace it |
22:02.26 | dlynes_office | chin1: you tried the other line port, right? |
22:02.32 | chin1 | no |
22:02.59 | dlynes_office | try it? |
22:05.07 | Symm | so umm how do i find out how much a pstn will charge me for connecting to them? |
22:05.20 | chin1 | that one doesn't work either |
22:05.20 | dlynes_office | Symm: you mean voip provider? |
22:05.22 | chin1 | i think its a setting |
22:05.27 | *** join/#asterisk doolph (n=doolph@200.75.204.169) |
22:05.31 | chin1 | there is no store for generic brand |
22:05.41 | dlynes_office | chin1: generic brand? |
22:05.57 | Symm | i guess, im still trying to work out the full topography of a US based voip system, from home |
22:05.58 | chin1 | its not vonage or anything locked in |
22:06.02 | doolph | what can I do if I get error on compilation --> [chan_zap.o] |
22:06.12 | Symm | ill just hit the books |
22:06.22 | Symm | i bought some asterisk book off amazon i just need to read |
22:06.28 | generalhan | ~book |
22:06.29 | jbot | extra, extra, read all about it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
22:06.37 | generalhan | yay jbot !!! |
22:06.59 | doolph | hello |
22:07.04 | generalhan | jbot: generalhan is THE MAN |
22:07.05 | jbot | okay, generalhan |
22:07.05 | kink0 | hmmmmmmmm |
22:07.05 | chin1 | dlynes_office: it works! |
22:07.06 | kink0 | ~h323 |
22:07.08 | jbot | extra, extra, read all about it, h323 is An ITU-T standard for packet-based multimedia communications systems. This standard defines the different multimedia entities that make up a multimedia system - Endpoint, Gateway, Multipoint Conferencing Unit (MCU), and Gatekeeper - and their interaction. This standard is used for many voice-over-IP applications, and is ... |
22:07.11 | generalhan | ~generalhan |
22:07.12 | jbot | i guess generalhan is THE MAN |
22:07.15 | generalhan | haha |
22:07.18 | doolph | how can I skip the that chan_zap |
22:07.27 | Dr-Linux | Juggie: awwwwwww you were right |
22:07.42 | dlynes_office | chin1: ? |
22:07.45 | chin1 | dlynes_office: i switched to trapezoid wave form sinusoid |
22:07.49 | kink0 | ~IXC |
22:07.50 | jbot | well, ixc is an interexchange carrier, also known as a long distance company, that transports calls between LATAs. Some examples are Sprint, Global Crossing, MCI, and AT&T. |
22:07.50 | dlynes_office | chin1: huh? |
22:08.05 | chin1 | dlynes_office: it says "Ring Waveform:" |
22:08.15 | chin1 | and has two options sinusoid and trapezoid |
22:08.15 | dlynes_office | chin1: what did you have it set to? |
22:08.16 | generalhan | ~Hmmhesays |
22:08.20 | chin1 | trapezoid |
22:08.25 | chin1 | it rang right away |
22:08.30 | dlynes_office | chin1: and now it's set to sinusoid? |
22:08.36 | chin1 | what ? |
22:08.37 | chin1 | no |
22:08.43 | chin1 | <PROTECTED> |
22:08.49 | dlynes_office | chin1: it was sinusoid and now it's trapezoid? |
22:08.53 | chin1 | yes! |
22:09.01 | dlynes_office | ok, you've got a weird phone then |
22:09.09 | dlynes_office | normally sinusoid should work just fine |
22:09.43 | Dr-Linux | there is fucking bug in RHEL 4 :@ :@ :@ :@ |
22:10.07 | dlynes_office | Dr-Linux: were you raised in the mountains? |
22:10.09 | kink0 | Dr-Linux, slackware !! heheehe is a block |
22:10.46 | Symm | http://www.gizmoproject.com/index.html |
22:10.46 | chin1 | dlynes_office: i knew i wasn't gonna take it back though you narrowed the answer down to 5 input feilds and you gave me the impression that the voltage was fine so the rest of hte options are really weird talking about frequency's and stuff i figured theyd be fine too the only white nad black answer wass the waveform |
22:11.15 | dlynes_office | chin1: yeah...that's just plain weird though...i've never had to change that setting |
22:11.16 | Dr-Linux | dlynes_home: what are you talking about |
22:11.22 | Dr-Linux | Juggie was right |
22:11.22 | dlynes_office | chin1: must be the cheap phone |
22:11.28 | chin1 | lol |
22:11.44 | dlynes_office | chin1: I always have expensive keysystems hooked up to them |
22:11.56 | kink0 | dlynes_office, may be due to A/D D/A conversion affected by bus clocking ? |
22:11.58 | dlynes_office | chin1: I've never used anything cheaper than a panasonic cordless hooked up to them |
22:12.23 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
22:12.23 | *** mode/#asterisk [+o russellb] by ChanServ |
22:12.30 | dlynes_office | kink0: well, i was thinking maybe the d/a converter in the cordless phone might only be a half wave converter, instead of full wave |
22:12.38 | *** part/#asterisk chin1 (n=Administ@c-68-84-57-212.hsd1.nj.comcast.net) |
22:12.47 | dlynes_office | kink0: that usually makes electronics cheaper |
22:13.01 | dlynes_office | kink0: but why bother, when you're only saving 40 or 50c? |
22:13.54 | kink0 | dlynes_office, yes, but that add a lot of armonic distortion, or you will need to reconvert the signal from an asimetric to simetric |
22:13.56 | dlynes_office | kink0: besides...i don't know why the phone would be dealing in AC to begin with |
22:14.08 | dlynes_office | kink0: the phone line is 70VDC, not 70VAC |
22:15.54 | dlynes_office | kink0: oh yeah..one other thing...no bus clockign...the pap2 is a wholly self-contained unit that gets plugged into the wall and the ethernet |
22:15.59 | kink0 | yes, I see, but signal is modulated like AC and added about 35 volts, so you always has full wave |
22:16.13 | Dr-Linux | kink0: my friend |
22:16.14 | Dr-Linux | = Parsing '/etc/asterisk/enum.conf': Found |
22:16.14 | Dr-Linux | Asterisk Ready. |
22:16.14 | Dr-Linux | *CLI> |
22:16.16 | Dr-Linux | :D |
22:16.29 | kink0 | Dr-Linux, did you used another gcc ? |
22:16.43 | dlynes_office | kink0: he's using redcrap linux |
22:16.50 | Dr-Linux | kink0: nope there was a bug in RHEL , Juggie helped me |
22:18.01 | kink0 | a question, any idea why IXC takes for her CDR start-time instead asnwer-time when they sends calls to my Asterisk ? |
22:18.22 | dlynes_office | who's her? |
22:18.39 | kink0 | dlynes_my peer, he used Cisco and IXC for accounting |
22:18.42 | dlynes_office | not to mention they? |
22:19.04 | dlynes_office | kink0: ah...I guess you must be Chinese |
22:19.25 | kink0 | he sends me h323 and the problem is her IXC takes start-time instead answer-time for CDR, so all calls are billed even BUSY or Congestion |
22:19.32 | dlynes_office | cause you're mixing him/her/they |
22:19.35 | kink0 | dlynes_office, hehehe |
22:19.58 | *** join/#asterisk rvhi (n=rv@66.175.65.89) |
22:20.07 | dlynes_office | ni shi zhongguo ren ma? |
22:20.10 | kink0 | ok, ... when they send :) |
22:21.12 | Dr-Linux | kink0: friend i'm going to home, i'll catch you from there |
22:21.30 | dlynes_office | Dr-Linux: lazy boy....always going home :) |
22:21.39 | kink0 | Dr-Linux, ok, take the bus , not the motocycle :) |
22:22.22 | Dr-Linux | dlynes_home: well i suppose to leave my office 5 hour ago .. anyway .. |
22:22.26 | Dr-Linux | bye |
22:22.30 | Dr-Linux | kink0: thanks friend |
22:22.31 | Dr-Linux | bye |
22:22.40 | Dr-Linux | /gone |
22:24.08 | dlynes_office | kink0: isn't that a good thing? |
22:24.12 | dlynes_office | kink0: you get more money then |
22:24.13 | dlynes_office | :) |
22:24.48 | kink0 | dlynes_office, no...no... that is not good !!! my local CDR is ok, my peer CDR is crazy |
22:25.01 | dlynes_office | hehe |
22:25.37 | kink0 | but I have other peers ussing Cisco to my Asterisk, and they has not problem, just this people who uses IXC has this problem |
22:25.54 | dlynes_office | sounds like a bug in ixc, whatever that is |
22:25.58 | *** join/#asterisk _4d4m_ (n=adam@62.69.102.99) |
22:26.01 | kink0 | all calls are computed with duration from start-time |
22:26.31 | kink0 | yes, that is what I think, I ask him if the problem is just with me, and they confirm they see this problem only with me :( |
22:26.35 | dlynes_office | ~wiki ixc |
22:26.36 | fholmes | how can I trace down potential incoming IAX communications? |
22:26.42 | dlynes_office | fholmes: iax debug |
22:26.59 | kink0 | I have asked to IXC softswitch ... no answer yet about the issue. |
22:27.06 | *** join/#asterisk RoyK (n=roy@122.80-203-78.nextgentel.com) |
22:27.23 | fholmes | Ok so how do I enable debugging? |
22:27.27 | dlynes_office | kink0: so you mean interexchange carrier, like wikipedia, or is ixc a piece of software? |
22:27.34 | dlynes_office | fholmes: iax debug |
22:27.45 | kink0 | is a piece of software, a softswitch for accounting |
22:27.46 | dlynes_office | fholmes: erm iax2 debug i mean |
22:27.46 | fholmes | nm. I read it wrong. Thanks |
22:27.59 | kink0 | sorry for confusion about interxchange or so |
22:28.00 | fholmes | I figured out the iax2 part atleast. :-) |
22:28.19 | dlynes_office | fholmes: iax2 no debug to turn it off |
22:29.05 | kink0 | anywise I think there some association or so , due I search for IXC in wiki and I got the site for interxchange who spokes about her softswitch software for voip accounting |
22:29.12 | fholmes | Ok, so I am registered with my IAX provider and can call out, but the calls are not coming back to me. I don't see anything come on the screen when I call my DID. Is there something I am missing? |
22:30.06 | dlynes_office | fholmes: is your iax connection defined as a peer, friend, or user? |
22:30.29 | fholmes | friend I believe. |
22:30.44 | dlynes_office | fholmes: so you don't know? |
22:30.50 | kink0 | dlynes_office, http://ixc.ua/index.php?MenuId=22 |
22:31.18 | fholmes | Well, I am just confused more than anything else. I have the register => line in there. However, down below in the iax.conf file I have type=friend |
22:32.09 | fholmes | I just noticed my username= was wrong down there. So maybe that was the issue. Let me reload real quick... |
22:32.15 | dlynes_office | fholmes: hrm...i've never had to do a register for iax |
22:33.02 | dlynes_office | kink0: it's incorrectly recording the calls when it's terminating on your end? |
22:33.06 | fholmes | So I don't have to have the register => line in the iax.conf file? |
22:33.20 | dlynes_office | fholmes: i'm guessing it would depend on your terminator |
22:33.31 | fholmes | register => account:pass@iax.provider.com |
22:33.35 | kink0 | dlynes_office, yes, that is the problem. |
22:34.09 | dlynes_office | kink0: i would check your end to make sure you're not using an answer anywhere in the dialplan for any calls coming in from them |
22:34.11 | kink0 | dlynes_office, her records shows all calls with duration ( even refused with Congestion or Busy calls ) |
22:34.20 | SkramX | fholmes: who are you using for termination? |
22:34.25 | fholmes | In the iax.conf file I don't exactly understand what he potential context= line might be used for? D |
22:34.28 | kink0 | dlynes_office, I did, I have not any Answer() |
22:34.29 | fholmes | Binefone |
22:34.36 | fholmes | Binfone* |
22:34.48 | dlynes_office | fholmes: the context= line is so that when they send you a call, that call will go into the context you've defined there, in the dialplan |
22:34.52 | kink0 | dlynes_office, also I have not this problem with other peers who sent me from similar Cisco |
22:34.54 | SkramX | hmm |
22:35.11 | dlynes_office | kink0: are they all going into the same incoming context? |
22:35.28 | kink0 | dlynes_office, yes , the same context for all |
22:35.42 | fholmes | dlynes_office: The call has to be passed to me first though right? Before the context comes into play. I do have a context setup for incoming calls in the extensions.conf file. |
22:36.05 | dlynes_office | kink0: then maybe their setup is broken...you might need to give them their own incoming context and set it up special somehow |
22:36.17 | dlynes_office | kink0: i.e. maybe their sip stack is broken |
22:36.30 | fholmes | SkramX: I can call out. I see the registration happening in the console..... |
22:36.31 | *** join/#asterisk xachen (i=justin@pdpc/supporter/student/xachen) |
22:36.35 | *** join/#asterisk crshman (n=chatzill@hacienda-heights-cuda2-68-71-5-62.lmdaca.adelphia.net) |
22:36.39 | kink0 | dlynes_office, I had try, but for some reason contexts inside my h323.conf are not working and use the general context |
22:37.02 | dlynes_office | fholmes: what's your context= line say in your iax context in your iax.conf file? |
22:37.12 | kink0 | dlynes_office, if I set a context for a peer in h323, is like ignored, that happens just with h323 |
22:37.13 | dlynes_office | kink0: oh...this is for h323...forgot |
22:37.24 | kink0 | yes, they used h323 |
22:37.34 | dlynes_office | kink0: i have no clue on that one....h323 is barely even supported in asterisk |
22:37.55 | fholmes | dlynes_office: Right now it says context=incoming-IAX |
22:37.57 | crshman | hi all in my logs i have this: "Allocating new SIP dialog for xxxxxxxxxxxxxxxxxxxxxxxxxx@127.0.0.1 - REGISTER (No RTP)" i am using broadvoice and i can't seem to get it to authenticate, any ideas? |
22:38.02 | kink0 | i know, I have a lot of troubles with h323, but as my peer is ussing, I am forced to support it |
22:38.08 | Symm | is there any need to learn RAGI, ruby on rails stuff.. or is there enough open source software available to set up a good voip system? |
22:38.10 | dlynes_office | kink0: but maybe they're expecting a specific q921 result code, and you're sending them the wrong one |
22:38.39 | kink0 | dlynes_office, I sent actually ISDN Cause 17 ( Busy ) |
22:38.39 | dlynes_office | kink0: you might end up having to hack your own changes into the h323 channel driver code |
22:39.23 | dlynes_office | fholmes: now do you have a context in your dialplan called [incoming-IAX]? |
22:39.38 | kink0 | dlynes_office, is more easy for me, I can change causes in the gateway ( my asterisk is connected to a 2N Stargate ) |
22:40.15 | dlynes_office | kink0: ah...cool |
22:40.36 | fholmes | dlynes_office: Yes, I do. Here is what I have in there: exten => _19995552342,1,Dial(SIP/1234) |
22:40.39 | dlynes_office | kink0: so they go into the gateway first, and hten into asterisk? |
22:41.03 | dlynes_office | fholmes: set verbose 6 at the cli |
22:41.04 | *** part/#asterisk Egonis (n=Egonis@207.245.14.10) |
22:41.06 | kink0 | dlynes_office, no, in reverse order, they call to my asterisk and then call is route to the GSM ussing the 2N |
22:41.19 | dlynes_office | fholmes: then make a call into your voip line, and see if it drops into asterisk |
22:41.44 | dlynes_office | kink0: ah...the 2N is an h323<=>GSM gateway? |
22:42.02 | fholmes | dlynes_office: No it is not going through. |
22:42.17 | dlynes_office | fholmes: check your /var/log/full log |
22:42.19 | kink0 | dlynes_office, is asterisk -> PRI -> GSM , also I am ussing SIP, and IAX is supported, no just h323 |
22:42.36 | *** join/#asterisk Talmage (n=Talmage@mychoice-fw.mychoice.cc) |
22:42.53 | dlynes_office | kink0: but between the 2N Stargate and Asterisk it's only h323? |
22:43.14 | kink0 | dlynes_office, nooo, there q931 signalling between them |
22:43.29 | dlynes_office | kink0: so it's a pri link then? |
22:43.36 | kink0 | h323 arrives to the asterisk, and do not pass to the 2N |
22:43.39 | Talmage | I have the pap2-na adpaters...and I am trying to use sip notify to restart the adapter...but it requires that the request be authenicated...how do I do so? |
22:43.48 | kink0 | yes, there a digium TE405 card there |
22:44.12 | dlynes_office | kink0: ah, and you've got one span connected to the 2N then, right? |
22:44.23 | kink0 | dlynes_office, right. |
22:44.28 | dlynes_office | ok |
22:45.00 | dlynes_office | kink0: and so the result code from the pri (2N) will get passed back via q.931 to the h323 link? |
22:45.04 | fholmes | dlynes_office: I don't see that log file anywhere. Is there another log file I need to look for? |
22:45.19 | kink0 | even code/decode is done by soft, I prefered Asterisk compared with gateway manufacturers voIP cards, so I use just a PRI interface from the gateway manufacturer. |
22:45.38 | dlynes_office | fholmes: edit your logger.conf file then, and make sure you've got a line: full => errors,warnings,notice,verbose,debug,dtmf |
22:45.49 | kink0 | dlynes_office, right !! the cause is passed, I have a translation table, but normally all causes are passed as is |
22:45.51 | dlynes_office | fholmes: then do a logger reload from the cli |
22:46.06 | dlynes_office | fholmes: then try your inbound call again |
22:46.11 | dlynes_office | fholmes: and then check the full log file |
22:46.35 | dlynes_office | kink0: find out from your ixc peer what causes they're expecting |
22:46.36 | *** join/#asterisk SmittyHalibut (n=msmith@adsl-69-239-168-105.dsl.snlo01.pacbell.net) |
22:46.45 | dlynes_office | kink0: they might be different from the cisco peeps |
22:47.35 | kink0 | dlynes_office, yes, that is what I have asked him today, but they really don't know, I will need to try with several until find what one will be ok, in the case that the problem is due to a not reconized isdn cause |
22:47.47 | SmittyHalibut | Question about the ZapTel drivers. All documentation I've been able to find says to download the ZapTel drivers via cvs.digium.com, but that hostname doesn't appear to resolve for me. Am I looking in the wrong place? |
22:48.05 | dlynes_office | kink0: they've got a complicated piece of hardware/software, and they don't know how to use it? |
22:48.12 | CunningPike | SmittyHalibut: Yes, you are - CVS was replaced by SVN a while ago |
22:48.13 | dlynes_office | kink0: that begs the question, why did they buy it? |
22:48.26 | kink0 | in the other hand, I must take care not cause missunderstanding for the actual well working cisco peers, since I am unable to set a context different for every one peer |
22:48.42 | dlynes_office | kink0: oh...why's that? |
22:48.52 | SmittyHalibut | /CunningPike: just svn.digium.com? |
22:48.58 | kink0 | dlynes_office, there question I dont ask :) I will never used what they are ussing |
22:49.06 | dlynes_office | kink0: why can't you set the ixc peer up on their own context? |
22:49.24 | kink0 | dlynes_office, because if I set differents context for every one peer in h323.conf appears to be completelly ingnored by Asterisk |
22:49.24 | Talmage | I have the pap2-na adpaters...and I am trying to use sip notify to restart the adapter...but it requires that the request be authenticated, how do I authenicate the request? |
22:49.43 | CunningPike | SmittyHalibut: If you want to run trunk (unstable), yes |
22:49.50 | dlynes_office | Talmage: go into yoru pap2-na adapter, and tell it not to use authentication |
22:49.53 | fholmes | dlynes_office: Nothing in there. Man, it has to be something with my provider. |
22:50.08 | CunningPike | SmittyHalibut: If you want stable, just go to ftp.digium.com and get the tarball |
22:50.18 | Talmage | dlynes_home Little worried about someone else sending my customer's sip notify messages. |
22:50.27 | dlynes_office | fholmes: well, if it was something on your end, i'm sure you'd see something about the call being sent into an invalid context |
22:50.50 | dlynes_office | Talmage: ah...thought this was for a residential setup |
22:50.51 | SmittyHalibut | CunningPike: That's exactly what I wanted to hear. Thank you. :) |
22:51.04 | dlynes_office | Talmage: i'm not sure how to do it, personally |
22:51.05 | CunningPike | SmittyHalibut: np |
22:51.13 | fholmes | dlynes_office: Is there anyway it could have anything to do with the codecs involved? GSM/alaw/ulaw etc? |
22:51.13 | dlynes_office | Talmage: i always use http to do it |
22:51.26 | Talmage | Well, some of them are natt'd |
22:51.27 | Talmage | etc |
22:51.33 | Talmage | They call and we find a configuration problem |
22:51.49 | dlynes_office | fholmes: if that was the case, you'd still see an error in your full log file |
22:52.01 | dlynes_office | Talmage: yeah...all but one of my customers are natted |
22:52.11 | dlynes_office | Talmage: i open up a port mapping on their router, though |
22:52.30 | fholmes | dlynes_office: Thanks for your help. |
22:52.44 | Talmage | Takes 10 minutes to tell them how to power cycle their ata |
22:52.55 | dlynes_office | unplug the power, plug it back in |
22:52.59 | dlynes_office | how difficult can it be? |
22:53.02 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
22:53.04 | *** join/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
22:53.10 | Talmage | ...they start unplugging things and the lights go out |
22:53.28 | dlynes_office | That's gotta be an american company :) |
22:54.20 | dlynes_office | even my dumbest customers aren't that bad |
22:55.24 | Talmage | Yeah... |
22:55.38 | Talmage | and a few of my customers are like 60 year old guys |
22:55.43 | Talmage | who hit on our female csr... |
22:55.46 | dlynes_office | most of mine are that old |
22:56.00 | Talmage | (just had to take a call from him) |
22:56.08 | dlynes_office | most of our customers are mining companies |
22:56.11 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
22:56.24 | *** join/#asterisk Trojan_Hors1 (n=root@220.226.22.191) |
22:56.49 | Trojan_Hors1 | hullo guys |
22:56.59 | dlynes_office | Trojan_Hors1: dood...don't irc as root |
22:57.14 | dlynes_office | Trojan_Hors1: unless you feel like getting hax0red |
22:57.47 | Trojan_Hors1 | thanks for advice..... i will switch soon |
22:58.22 | dlynes_office | heh...on efnet, they won't even let you join any channels if you're irc'ing as root |
22:58.43 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
22:58.44 | Talmage | dlynes_office well thanks, will try asking my question tommorrow |
22:59.17 | Trojan_Hors1 | i need to know the largest user base a distributed asterisk servers hav been in production use |
22:59.20 | dlynes_office | Talmage: well, i'd read up on the sip rfc |
22:59.29 | dlynes_office | Talmage: it might help you figure out how to do it |
22:59.47 | russellb | Trojan_Hors1: i know of installations with 10's of thousands of users ... |
22:59.56 | SmittyHalibut | CunningPike: You rule. I've got them, compiled them, and am continuing my day. Thanks again. :) |
23:00.17 | CunningPike | SmittyHalibut: Don't thank me - thank Mark Spencer ;) |
23:00.32 | CunningPike | SmittyHalibut: Actually, thank russellb and the crew |
23:00.36 | dlynes_office | Talmage: are you in the US? |
23:00.37 | Trojan_Hors1 | russellb, wats it ? |
23:01.08 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.18.81.Dial1.SanJose1.Level3.net) |
23:01.26 | dlynes_office | or anyone else in the US that wouldn't mind calling a phone number for me? |
23:02.11 | Mw3 | hm, does linksys pap2 support t.38? |
23:02.17 | dlynes_office | Mw3: yes |
23:02.42 | *** join/#asterisk chaoscon (n=ph33r@smartserv/ceo/chaoscon) |
23:02.43 | dlynes_office | Mw3: no idea how to set it up though |
23:02.48 | Mw3 | any special configuration needed |
23:02.49 | Mw3 | ? |
23:02.51 | Mw3 | oh, ok :) |
23:03.06 | *** join/#asterisk Seyr (n=Seyr@cpe-67-10-139-141.houston.res.rr.com) |
23:03.29 | dlynes_office | Mw3: try checking the sipura and/or linksys users' group on voxilla |
23:03.30 | kink0 | see you later !! good night guys |
23:03.45 | Mw3 | dlynes_office: i'll do that. thank you |
23:03.46 | dlynes_office | seeya kinky |
23:04.25 | Trojan_Hors1 | okey for a userbase of 3000 wats d optimal number of asterisk servers [assuming 30 PSTN lines] |
23:04.28 | *** part/#asterisk Seyr (n=Seyr@cpe-67-10-139-141.houston.res.rr.com) |
23:04.41 | mitcheloc | 1 server per user |
23:04.46 | dlynes_office | heh |
23:04.52 | Bert- | Hmm |
23:04.53 | Trojan_Hors1 | errr |
23:05.04 | mitcheloc | (it's a joke) |
23:05.11 | mitcheloc | 30 lines is enough for 3000 users? |
23:05.34 | dlynes_office | mitcheloc: why not? most of the time they're not going to be calling out, only calling each other |
23:05.35 | Trojan_Hors1 | ya..... d main business is on PC to PC |
23:05.36 | Trojan_Hors1 | softphones |
23:05.38 | Bert- | I have a VoIP account which use MGCP protocol. Can asterisk handle it like a simple PSTN line ? |
23:06.02 | dlynes_office | Bert-: asterisk has an mgcp channel driver |
23:06.09 | Bert- | ok |
23:06.25 | mitcheloc | Trojan_Hors1: you could do that with ser then, and only like 2-3 boxes i bet |
23:06.26 | dlynes_office | Bert-: so you can use the Dial() application for that channel just like any other cahnnel |
23:06.39 | Bert- | because I'm unable to find a linux softphone which use MGCP : |
23:06.40 | Bert- | :( |
23:06.59 | dlynes_office | Bert-: i think you're going to be hard pressed to find an mgcp softphone for any operating system |
23:07.10 | Bert- | I4ve found one for win |
23:07.20 | Bert- | but I use Linux ... |
23:07.33 | dlynes_office | Bert-: qemu? |
23:07.47 | Bert- | hmm never tested |
23:07.49 | Bert- | let me try |
23:08.07 | dlynes_office | Bert-: http://sf.net/projects/qemu/ |
23:08.26 | Bert- | thx |
23:08.43 | Trojan_Hors1 | mitcheloc, okey 2-3 boxes arnt a big deal...... but are ye sure i expect an avg of 30-40 concurrent cnxns and a high of 150-200 |
23:09.01 | dlynes_office | Bert-: erm...hold on...that's not the url |
23:09.09 | Bert- | I see ... :) |
23:09.26 | Talmage | dlynes_office yes, sorry on tech support call. |
23:09.37 | *** join/#asterisk pdavid (n=chatzill@adsl-068-209-191-127.sip.mob.bellsouth.net) |
23:09.53 | SmittyHalibut | CunningPike: I've updated the Wiki with the new process, FTP instead of CVS. Thanks again. And, thanks to Russellb and Mark Spencer! :) |
23:10.04 | pdavid | hi all! could anyone suggest a good voip provider for a small business in southern US |
23:10.05 | pdavid | ? |
23:10.14 | dlynes_office | Talmage: yeah...was just looking for someone from the us that'd be able to call one of my dids for me...i've got a line quality issue that only appears on calls from us users |
23:10.14 | *** part/#asterisk SmittyHalibut (n=msmith@adsl-69-239-168-105.dsl.snlo01.pacbell.net) |
23:10.16 | mitcheloc | Trojan_Hors1: i'm *not* speaking from experience, but what i understand is nothing goes through the ser machine it'self, so it proxies everything, should be pretty easy |
23:10.29 | mitcheloc | ftp = new process??? |
23:10.29 | dlynes_office | Talmage: i'm just curious how widespread the problem is |
23:10.45 | Talmage | is it intl? |
23:10.57 | dlynes_office | Talmage: not really...Canada |
23:11.13 | mitcheloc | Trojan_Hors1: i don't use it, suggest you check #ser |
23:11.25 | Talmage | I am already getting yelled at for making too many intl calls |
23:11.30 | dlynes_office | lol |
23:11.31 | Talmage | then again...I could just empty the cdr db |
23:11.31 | Trojan_Hors1 | okey thanks for the help |
23:11.40 | dlynes_office | hahah |
23:11.44 | dlynes_office | don't worry about it then |
23:11.51 | pdavid | i have been considering voicepulse, but wanted some opinions on it if anyone had any... |
23:12.25 | generalhan | pdavid: i have used VP in the past ... i dropped them hardcore and am paying more for a PRI T-1 and its sooooo worth the price |
23:12.45 | dlynes_office | Bert-: http://fabrice.bellard.free.fr/qemu/ |
23:12.46 | generalhan | VP had been dropping about 80% of my packets to them and they refused to do any testing on there end to help me figure out why. |
23:12.52 | pdavid | generalhan: what type of issues caused you to drop them? |
23:12.56 | pdavid | ahh |
23:13.02 | generalhan | 80% is A LOT ! |
23:13.12 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
23:13.14 | pdavid | that is a lot |
23:13.43 | generalhan | i understand that CAN happen but i may have stayed with them if they took more time to help me figure out where the routing issue was taking place |
23:13.50 | dlynes_office | generalhan: i hope you were getting 0.00000001c/min for your calls? |
23:14.09 | generalhan | yea right ... iwas credited ANYTHING for my loss |
23:14.11 | nahirean | general: how can you be sure VP is dropping the packets? how do you know which router is actually losing them |
23:14.19 | generalhan | s/iwas/i wasnt |
23:14.43 | pdavid | yes, did you ever resolve the packet loss? |
23:15.13 | nahirean | general: any router along the way could be tossing those packets like a salad. |
23:15.17 | rpm | http://pastebin.ca/62749 - does this logic make sense to anyone? i want to be able to transfer a caller to an extension and place them on hold via an external source. and when they are taken off hold it call me back |
23:15.19 | generalhan | nahirean: yea ... i had been doing pingplotter tests for weeks straight .. i knew which server on their end was starting the packet loss but they kept calling me a liar ! lol |
23:16.07 | generalhan | and let me clear up something real quick .. i KNOW that its not all their fault .. but the reason that i was soo dissatisied was because of their apathy about my entire situation |
23:16.29 | nahirean | general: i have an TF number with VP and i kid you not, i capped out my bandwidth using a binary newsreader, and the quality was still awesome.. seriously, ive not had any issues with them |
23:16.43 | pdavid | well, does anyone have any other thoughts/recommendations on a provider? |
23:16.44 | generalhan | nahirean |
23:16.48 | Talmage | teliax |
23:17.05 | pdavid | its like a primordial voip ooze sifting through all the providers |
23:17.35 | pdavid | i was just looking at teliax |
23:17.59 | generalhan | nahirean: yea ... i know lots of people that have used them and not had any issures ... but when i call a provider of mine and tell them that i cant use their service because of XYZ i expect them to show some concern and/or sympathy about my issues ... they were just unwilling to except any responsability |
23:18.01 | pdavid | you know, its a pet peeve of mine when someone says "Unlimited***", then the footnote reads: *** Softcap of 2500 minutes |
23:18.27 | pdavid | 2500!=inf |
23:18.36 | dlynes_office | pdavid: vonage swears up and down that they've got unlimited calls |
23:18.47 | pdavid | it's just shady practice |
23:18.56 | pdavid | why not just say: 2500 minutes |
23:18.57 | dlynes_office | pdavid: but apparently if you exceed what they deem to be within normal limits, you get flagged |
23:19.09 | *** join/#asterisk JoseBravo (n=jdbravo@200.24.110.91) |
23:19.13 | pdavid | yeah, and/or backcharged like some people complaining about broadvoice.com |
23:19.26 | pdavid | at least be up front about what you are offering |
23:19.36 | generalhan | pdavid: what i would do is get some form of ping plotter to trace 24 hours of data to their server ... if its good then i would go with VP .. i had their wholesale minutes and i had 30k minutes and paid dirt for it |
23:20.02 | pdavid | generalhan: thanks, i think i will try that, and let it run all night/day tomorrow |
23:20.09 | pdavid | see how it pans out |
23:20.12 | generalhan | they ARE really really cheap with their wholesale minutes ... if i could have kept a constant connection to them i prolly wouldnt have left |
23:20.31 | pdavid | i don't mind paying a little extra for the service |
23:20.36 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
23:20.47 | pdavid | i was going to use their connect plan, to use my * |
23:21.05 | pdavid | oddly, their calls to canada were more expensive than calling australia... |
23:21.11 | generalhan | pdavid: 66.234.228.134 is the server they were using when i had it ... you can test on that one .. otherwise just call them and ask ... if you tell them what you need it for they will be happy to give you the ip |
23:21.21 | *** join/#asterisk Drew___ (n=foo@zux221-156-100.adsl.green.ch) |
23:21.29 | dlynes_office | pdavid: it's more expensive for me to call the US, than to call China :) |
23:21.37 | dlynes_office | pdavid: and that's from Canada |
23:21.43 | pdavid | freakin yikes |
23:22.11 | dlynes_office | pdavid: my rates to Canada though are less than half of China |
23:22.32 | mitcheloc | it's free for me to call anywhere in the US with skype! |
23:22.36 | pdavid | dlyines_office: who are you with? |
23:22.43 | dlynes_office | pdavid: I buy wholesale |
23:22.49 | pdavid | ahh |
23:23.00 | pdavid | so any other providers i should consider? |
23:23.04 | dlynes_office | pdavid: it wouldn't matter who i buy from because they don't sell retail |
23:23.16 | *** join/#asterisk ToTo (n=ToTo@host212-109.pool8258.interbusiness.it) |
23:23.28 | pdavid | i was considering BV, but have read many bad reviews of late that have sort of turned me sour |
23:24.02 | Juggie | whats the best place for wholesale iax/sip minutes? |
23:24.09 | *** join/#asterisk P-NuT (n=P-NuT@fw.office.unitedip.net.au) |
23:24.19 | dlynes_office | pdavid: try www.calltermination.com |
23:24.36 | dlynes_office | Juggie: you might want to qualify that |
23:24.58 | dlynes_office | Juggie: what do you consider to be 'best'? cheapest? best call quality? or a nice easy medium? |
23:25.04 | pdavid | dlynes_office: thanks, checking it out now |
23:26.35 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
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23:26.53 | *** part/#asterisk ChewyNet9 (n=plugci@adsl-61-111-148.sdf.bellsouth.net) |
23:27.16 | Juggie | dlynes_home, best of both worlds |
23:27.47 | dlynes_office | Juggie: well, i've had pretty good luck with Five9sNetwork |
23:28.09 | dlynes_office | Juggie: they're relatively stable, call quality is good, and they're cheap |
23:28.33 | dlynes_office | Juggie: the only issue i've had with them, is sometimes I can't make calls to certain destinations |
23:28.43 | dlynes_office | Juggie: usually for my Indian white routes |
23:29.21 | *** part/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
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23:33.35 | Drew___ | how would i get asterisk to simply connect a sip phone with a zap channel - i.e. without dialing anything? |
23:34.07 | *** join/#asterisk Zodiacal (i=hehehe@bdsl.66.14.242.199.gte.net) |
23:34.08 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
23:34.14 | JoseBravo | How I can check if my asterisk is connected to astbill Mysql db? |
23:34.17 | *** join/#asterisk mogorman (i=ejabberd@68.62.237.103) |
23:34.30 | Zodiacal | anyone setup a cisco 7914 before? with sccp? |
23:34.48 | Zodiacal | do i need to specific two devices in my sccp.conf? |
23:35.03 | Zodiacal | one for 7960 and one for 7914? |
23:35.13 | Zodiacal | and a line for each? |
23:35.55 | Zodiacal | to specific = specificly |
23:38.36 | *** part/#asterisk doolph (n=doolph@200.75.204.169) |
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23:46.55 | Umaro | Hi.. anyone here from digium that I can talk to about my g729 license order? |
23:47.02 | mogorman | whats up |
23:48.14 | Umaro | mogorman, do you work for digium? |
23:48.23 | mogorman | that i do |
23:54.31 | brimstone | Umaro, what's up? |
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